RE: [Asterisk-Users] Trying to find good VOIP provider.
I don't know if you keep your eye on the -biz list or not. But you should if you don't. Plainvoip.com just anounced last weekend they are offering blended US48/Canada Termination @ .007 w/ free Toll Free termination. They support IAX/SIP w/ all major codecs including g723 and g729. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lachek Butalek Sent: Thursday, June 15, 2006 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Nikolay Pavlov Subject: Re: [Asterisk-Users] Trying to find good VOIP provider. Voxee.com supports both SIP and IAX2, as well as GSM, iLBC, uLAW, aLAW and G729 codecs. Their rates to international locations are based on 6/30 billing, and vary a lot from location to location. You can view their rates here: http://www.voxee.com/rates.xls You may also want to look at VoipStunt.com, although I believe they are SIP-only. Their rates are very low, though - perhaps you could use them as a complement to another provider for normally high-priced locations. On 6/15/06, Nikolay Pavlov [EMAIL PROTECTED] wrote: Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- = = Best regards, Nikolay Pavlov. = = ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Cell gateway for T-Mobile US??
Typically yes, as long as you can get power for them compatible with ours. Tmobile is GSM. Well only GSM. They don't do anything else. You can check the WIKI I have found a few smaller ones that will probably work but don't remember what they are except that I found them there. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz, Steven Sent: Monday, June 12, 2006 9:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cell gateway for T-Mobile US?? Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the asterisk side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 passthrough/middleman
There are a number of ways to do this. You can use your Dual T1 to do what you want. You bring your CO T1 line to the card which gives your inbound and local outbound. Your 2nd T1 can go to your legacy PBX. You just have to setup your dial plan accordingly to route the calls the way they need to go. I have the same situation @ my office. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Friday, June 09, 2006 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] T1 passthrough/middleman Is it possible to act as a middle man on a T1 line? My installation currently has an aging Inter-Tel Axxess box with a T1 coming in (16 in, 8 out). Rather than adding and replacing phones and cards as they die, I would like to slowly migrate to a asterisk SIP installation. I want to take the incoming T1 line, use any available outgoing lines for outgoing SIP, intercept any incoming lines and either send them off to a SIP line or pass them through to other T1 line (going to the Axxess box), and finally take in outgoing calls from the Inter-Tel box and either send them to SIP or send them to the outside T1 line. How will a dual T1 card be set up in this situation? Would it be easier to use an FXO channel bank (or card) and connect analog lines to the FXS analog lines on the Inter-Tel box? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Quad T1 Card
Asterisk Hater.. :) Sorry matt couldn't resist.. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Wednesday, June 07, 2006 11:20 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quad T1 Card Hello, I have done a lot of testing on both the Digium TE406P and the Sangoma a104d and was involved in debugging both of them with Digium and Sangoma in their early releases. Since we are on a Digium-owned list right now and I don't want to be branded an enemy of Asterisk again for suggesting that you might consider buying a non-Digium product, I will mention right up front that a large portion of your purchase price from buying a Digium card will go toward keeping Asterisk development going, in fact it is how Digium makes most of their money and allows them to have dozens of programmers working full time on Asterisk. Sangoma does contribute to the Asterisk codebase, but buying a Sangoma card will not help the owner of Asterisk further improve their product at all. Now on to my recommendation. As I mentioned we have had both the Digium and Sangoma echo-cancellation cards in production for over 6 months on heavy load Asterisk servers running both 1.2.X Asterisk. Both had initial problems with drivers with the Sangoma side being fixed within a couple weeks and the Digium side being fixed by having to manually disable the hardware DTMF detection in the wct4xxp.c driver code every time I upgrade zaptel. Both of the cards do a good job at removing echo from our calls, and they both have a fairly equal effect of reducing the overall load on your system(10-20%). So performance-wise in our tests in our environment they are pretty much the same. As for the technical specs on the echo-cancellation modules used, the Sangoma card uses an Octastic chipset that is highly configurable and is one of the best telecom echo-cancellation chipsets in the industry. Is has a configurable tail length and is capable of dynamically being turned on and off as needed by it's firmware. The Digium card uses an Oki chipset that has a smaller echo tail length and is hard-coded into the firmware so you cannot change it. The other differences are just the usual differences between Digium and Sangoma cards: Digium - ready to go just loading zaptel and Asteirsk, Sangoma - must load wanpipe drivers and configure each span before using, also must recompile zaptel after installing/upgrading wanpipe driver Digium - 2 year warranty, Sangoma 5 year warranty Digium - has motherboard incompatibility list, Sangoma - guarantees functionality with all modern PCI-compliant motherboards Hope that helps, MATT--- On 6/7/06, Rich Adamson [EMAIL PROTECTED] wrote: Sean Cook wrote: One of the primary differences between the two cards is the Sangoma h/w echo canceler handles more cases of echo then do the Digium cards. Whether you need that additional coverage is 100% dependent on your specific implementation (eg, your T1/PRI provider), and not on what the list thinks about the two products. Since there are no affordable tools to truly quantify echo for each specific implementation, as a pbx engineer your toolkit should probably include both cards. Sort of like try the less expensive card and if it doesn't address your echo issues, then try the more expensive one. No offense but isn't that like saying Don't take what the list has to say about your purchase... instead you should guess and hope you get the right answer... but if you don't, gamble again and buy two cards? The list cannot guess at what level of echo you are going to incur, therefore there is no way for anyone to accurately tell you how to address issues. Both cards are quality products, but with slightly different operational characteristics. If you can't afford to purchase both cards, then a safe bet is to simply purchase the Sangoma card since it can address more echo issues then the Digium card. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be
RE: [Asterisk-Users] Block access to [EMAIL PROTECTED]
In your sip.conf or iax.conf you need to change the default context to something that will not interact with your main dialplan. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d:+1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pietro U Sent: Wednesday, June 07, 2006 1:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Block access to [EMAIL PROTECTED] i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this? This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Prices of g729 codec
Now why would you want to go and not support Digium and the community for their hard work to produce a quality product? $10 isnt that much for using the licenses.. If you take into consideration of how much it COULD cost to purchase something like this based on circuits it would be insane. Don't cheat digium out of money.. pay the $10 per license. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frédéric Marti Sent: Friday, June 02, 2006 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Prices of g729 codec You can also build G729 codec by urself via Intel IPP. Regards === Do you know if they are compatible with Digium's codecs? Like this exemple: 2 Asterisk linked via IAX2 , 1 with Intel's codec and 1 with Digium's codec. Regards Fred === -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Friday, June 02, 2006 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Prices of g729 codec Hi, does anyone know the prices for g729 codecs from Digium? I sent an email a while ago to them but haven't got any response so far. Prices are per unit or volume? Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Prices of g729 codec
eh? I try.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, June 02, 2006 12:13 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Prices of g729 codec On Friday 02 June 2006 11:39, Lee Howard wrote: Don't cheat digium out of money.. pay the $10 per license. Yes, be a good colonist and don't dump any more tea into the harbor. Oh please. Brian's got the reasoning for paying for the license entirely wrong but at least his heart's in the right place. The Intel g729 code is licensed for educational use ONLY. Commercial use is forbidden without paying the patent holder. $10 a port won't break the bank of any business with a shred of a hope of a chance of surviving, and you stay legitimate. Try buying a legit g729 license from the patent holder if you're a home user or small business wanting to transcode g729. They only want to license hundreds of instances at a time, if not thousands. Digium negotiated a pretty damn good license fee so that they could offer the codec and sell it in onesie-twosie quantities to little guys like us at an affordable price. It's $10 per simultaneous transcode, Lee. It's not per month or even per-year. It's a one time fee. If you're a major carrier, chances are you aren't transcoding g729 on too many channels on PCs anyway, instead relying on the already-paid-for, already-legit g729 codecs on your termination equipment (Cisco, Lucent, etc.). In that case, spending $100 or even $1000 on g729 licenses (scaled for your needs of course) is a paltry sum compared to the equipment you have in place already to run the rest of the VOIP end of the business. be a good colonist indeed. You've got your head so far up your arse you've entered a new and entirely intestinally-based existence. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Prices of g729 codec
I don't see a problem with monetary reward for hard work. If it wasn't for Mark, the Digium Team, and the community of developers you wouldn't have what you have. I am thankful for open source projects and support in anyway I can.. Money or otherwise. So say I'm brainwashed or employed either way.. I support and stand behind Digium 100% brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Friday, June 02, 2006 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Prices of g729 codec On Friday 02 June 2006 11:39, Lee Howard wrote: Don't cheat digium out of money.. pay the $10 per license. Yes, be a good colonist and don't dump any more tea into the harbor. Oh please. Brian's got the reasoning for paying for the license entirely wrong but at least his heart's in the right place. It's $10 per simultaneous transcode, Lee. It's not per month or even per-year. It's a one time fee. be a good colonist indeed. You've got your head so far up your arse you've entered a new and entirely intestinally-based existence. I'm not advocating illegal activity. I am, however, mocking the the zealousness behind the reasoning, as you put it. The GPL very clearly defines the method of expressing deserved gratitude, and it is not in monetary support. Those who would like to extend the requisite gratitude further have motives, are brainwashed, or are employed by Digium. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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 crashes at startup
run asterisk with asterisk -c and see if it gives anymore information. You can also get it to produce a core dump and see if it gives you anymore information. brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, May 31, 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk crashes at startup Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP provider for Turkey from India with Asterisk
Plainvoip has a very good A-Z and I have found they are fairly inexpensive. They also offer TollFree orig and some local dids. www.plainvoip.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Friday, May 26, 2006 9:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP provider for Turkey from India with Asterisk Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to Turkey country from India. With VoIPJET, I am unable to make call to Turkey and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India. Looking forward for your response. ThanksRegards, Chandramouli Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] US telco lingo
Well we try.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Pobanz Sent: Thursday, May 25, 2006 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] US telco lingo Brian C. Fertig wrote: I think dude was trying to be a smart ass or show us his experience in telecom.. :) At least he knows the pinout for a T1.. I have been properly put in my place by you and many others.. :) After rereading the original post, I don't believe it has anything to do with jacks. I use to work for a local phone company where we regularly did T1 installs and the only 48 we used was part of a rj48 jack. Thanks, for not letting anything foolish get through!!! :) Don Pobanz -Original Message-n to a non-US dummy the following phrases I have What is US48? I assume by US48 they mean RJ48 which is a 8 conductor modular jack with signal from the phone company on 12 and signal to the phone company on 45. Don Pobanz You are kidding right??? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] FW: [isp-clec] Treasury disconnects tax on long-distance calls - with refunds
FYI Brian Fertig Treasury disconnects tax on long-distance calls WASHINGTON (MarketWatch) - The brief Spanish-American War ended more than a century ago, but not the federal tax assessed to fund the victory. Until now. On Thursday, the U.S. Treasury said it would stop collecting the 3% federal excise tax on long-distance calls, a fee originally assessed in 1898. The government also said it will issue refunds requested by consumers and businesses that paid the fee over the past three years. Taxpayers will be able to request refunds when they file 2006 tax returns in early 2007. The Treasury also said the Justice Department would cease litigation in support of the tax after a handful of federal appeals courts ruled the fee illegal in decisions rendered within the past year. The most recent loss in federal court occurred earlier this month. The Federal Appeals courts have spoken across the board, Treasury Secretary John Snow said in a statement. It's time to 'disconnect' this tax and put it on the permanent 'do not call' list. The tax, which generates more than $6 billion annually, has survived repeated efforts to eliminate it, most recently in 2000, when President Bill Clinton vetoed a larger bill that included a repeal of the excise fee. Bills aimed at ending the tax have circulated every year since. For decades, long-distance companies such as ATT Inc. have been required to collect the excise fee from customers and pass it on to the federal government. Yet some large corporations such as Hewlett Packard successfully sued to get rid of the tax, claiming it was illegal. Others have won large refunds from the IRS. The excise tax works out to $1.50 per every $50 in long-distance calls, not a particularly large sum for consumers. Yet for a business that spends, say, $10,000 a month on long-distance calls, the tax would equal $300 a month or $3,600 a year. If the tax remained in place over the next decade, it would have generated about $67 billion for the federal coffers, a congressional panel estimates. Altogether, the excise has raised more than $300 billion in its entire existence, the Congressional Research Service found. The excise fee was originally established in 1898 on long distance because phones were considered a luxury and only the wealthiest Americans could afford service. These days, the tax affects all consumers directly or indirectly, no matter what their annual income. In announcing his decision, Treasury Secretary Snow also called on Congress to eliminate federal taxes on local phone calls. That tax is separate from the long-distance fee. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] US telco lingo
I think dude was trying to be a smart ass or show us his experience in telecom.. :) At least he knows the pinout for a T1.. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, May 24, 2006 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] US telco lingo -Original Message-n to a non-US dummy the following phrases I have What is US48? I assume by US48 they mean RJ48 which is a 8 conductor modular jack with signal from the phone company on 12 and signal to the phone company on 45. Don Pobanz You are kidding right??? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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 in php-asmanager.php
You will need to modify your php.ini file to allow it to run longer. Normally if you exceed 30 seconds there is something majorly wrong with your app. Look for the following tag: max_execution_time Change it to 30sec. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo Souza Queiroz Sent: Tuesday, May 23, 2006 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem in php-asmanager.php Friends, I have message error in module Extensions in Asterisk : Fatal error: Maximum execution time of 30 seconds exceeded in /var/www/html/admin/common/php-asmanager.php on line 169 Do you know this message error ? Thank´s Gustavo Queiroz - Rio de Janeiro - Brasil This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Now that Nufone is dead...
I just got a DID from www.plainvoip.com the cost is $2.00 a month and 2c incoming. They also port TF w/ LOA. Brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Tuesday, May 23, 2006 10:49 AM To: Asterisk Subject: [Asterisk-Users] Now that Nufone is dead... Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Diverse servers
For your configuration to be like this RRDNS and Realtime. I believe someone made a patch for realtime to work correctly with RRDNS you would have to check the wiki or mantis to find it. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, May 17, 2006 9:51 AM To: Asterisk-users@lists.digium.com Subject: [Asterisk-Users] Diverse servers I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system such that in event of failure, childAsterisk boxes, phones, ATAs, etc. can register to either box. I can handle the child's configuration, but how do I have it setup on the Asterisk boxes? I'm not exactly sure I explained this right, but hopefully someone can get what I'm talking about and ask further questions of me. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet down?
I use Plainvoip.. And I know a lot of the community does.. Rates are inexpensive and quality is excellent. brian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.com Sent: Tuesday, May 09, 2006 3:40 PM To: asterisk-users@lists.digium.com Subject: SPAM [Asterisk-Users] voipjet down? Somebody know if they are down? Let me know, Julius C. Barber [EMAIL PROTECTED] www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet down?
I will agree 3.9c is quite expensive for termination. Most providers hover around the 1 to 2c mark. 3.9c is just a way for them to cover all of their overhead. I have found a lot of providers even at 1c can be very stable and offer good services. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, May 10, 2006 10:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] voipjet down? I cant imagine anyone using voipjet as their only or main provider. And I'll say again, 3.9 cents for an ITSP is the most expensive I have found. Business grade termination is typically much less than that with top notch companies like https://www.nexvortex.com/ at 2.5c. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, May 10, 2006 5:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-Users] voipjet down? And again I'll say... calleveryone.com for all your RELIABLE termination needs. And again... don't go by the rates on the page... those are the end-user rates... call them for wholesale rates.. they will be competitive to voipjet, and you get phone support and quick response time. Come on guys... if you are still using VoipJet, you don't care about your companies termination. On 5/9/06, Wes Baehr [EMAIL PROTECTED] wrote: Even stranger is when calls (to the same server) work from one asterisk server account, but fail from another asterisk server account. Sometimes changing the server helps, sometimes it doesn't. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 Fax: 330.882.0455 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, May 09, 2006 4:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] voipjet down? I havebent been able to call out in weeks and nobody returns emails to [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.com Sent: Tuesday, May 09, 2006 12:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voipjet down? Somebody know if they are down? Let me know, Julius C. Barber [EMAIL PROTECTED] www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 5/8/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 5/8/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipjet down?
ok then.. Where are their wholesale prices? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, May 10, 2006 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet down? Kerry, Do you have a reading problem? Both times that I have tried to help people out by suggesting a company I have personally used and have had good luck with, you reply and say that the rates are horrible. If you would read my e-mails you would see that the 3.9 cents is NOT for wholesale termination. If you want someone would will give cheap termination to end users, go use voipjet or whatever you want. If, on the other hand, you want some reliable cheap wholesale termination, go check out voipjet. On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote: I cant imagine anyone using voipjet as their only or main provider. And I'll say again, 3.9 cents for an ITSP is the most expensive I have found. Business grade termination is typically much less than that with top notch companies like https://www.nexvortex.com/ at 2.5c. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, May 10, 2006 5:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-Users] voipjet down? And again I'll say... calleveryone.com for all your RELIABLE termination needs. And again... don't go by the rates on the page... those are the end-user rates... call them for wholesale rates.. they will be competitive to voipjet, and you get phone support and quick response time. Come on guys... if you are still using VoipJet, you don't care about your companies termination. On 5/9/06, Wes Baehr [EMAIL PROTECTED] wrote: Even stranger is when calls (to the same server) work from one asterisk server account, but fail from another asterisk server account. Sometimes changing the server helps, sometimes it doesn't. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 Fax: 330.882.0455 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, May 09, 2006 4:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] voipjet down? I havebent been able to call out in weeks and nobody returns emails to [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.com Sent: Tuesday, May 09, 2006 12:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voipjet down? Somebody know if they are down? Let me know, Julius C. Barber [EMAIL PROTECTED] www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 5/8/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 5/8/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To
RE: [Asterisk-Users] Hi...Please help me
Chandra, In all honesty if they are proprietary and you want to use them you will need a FXO card. Alternatively there are a few good termination providers out there that are inexpensive. The top 3 most inexpensive that come to mind are: Plainvoip - http://www.plainvoip.com Domestic starting at 1.1c VoipJet - http://www.voipjet.com Domestic starting at 1.3c NuFone - http://www.nufone.net Domestic starting at 2c (I believe) Anyone of these providers can supply you with USA and also international dialing. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Monday, May 08, 2006 8:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hi...Please help me Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called Voice Finder AP 200 and the below values: Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz (These values are dummy values) Currently we are making US calls using VoIP provided by Vebtel. Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this? I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel? Waiting for your quick response. Thank you. Regards, Chandra. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] FW: NuFone Update: DIDs (Correction)
Well I know from personal experience that NuFone is working on a solution for its customers as fast as it can. I know they found an alternate termination provider and are working to have a solution for the TF and Local DID's he currently has on his platform. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wes Baehr Sent: Wednesday, April 19, 2006 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'Jeremy McNamara' Subject: RE: [Asterisk-Users] FW: NuFone Update: DIDs (Correction) Correction to my original post: My service has not been interrupted, as far as I know, and probably won't be. But, it probably wouldn't be a bad idea to have another provider in case Telesthetic does decide to cut NuFone off. (Although an agreement supposedly has been reached.) NuFone is still one of the best providers I've used so far, with great support (and prices). (And my toll-free service hasn't been interrupted either). -- Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wes Baehr Sent: Tuesday, April 18, 2006 3:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] FW: NuFone Update: DIDs Well this is disappointing. Time to find somebody else... -- Wes -Original Message- From: NuFone Operations [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 18, 2006 3:44 PM To: [EMAIL PROTECTED] Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate our services. We were only informed of such decisions on Thursday Afternoon, April 13th, 2006, which as anyone knows is abolutely not enough time to solve the complex matters at hand. Initially, we were hopeful that we could work out an acceptable deal, at the very least, to allow us enough time to move your telephone numbers to another carrier without much, if any service interruption. Sadly, this no longer seems to be the case. We are currently working with another carrier to host your Toll-Free numbers. However, you may want to consider submiting a Number Portability request to another carrier, to avoid any service outages caused by Telesthetic's threat to terminate our service. We can always port your number back to our service at later date, at no cost to you. If you would like to keep your Michigan telephone number you will need to contact Telesthetic directly at 248-724-0600 to determine if they will provide you service or not. We are going to do everything we can to survive yet another failed business partnership. We do not intend to give up. We can and will prevail through yet another time of great challenge. We will update you as soon as we have more information. Thank you for your continued support and words of encuragement. The NuFone Network http://www.nufone.net/ -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth 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] Little OT.. SER Question
Anyone with SER knowledge could you point me in a direction to setup SER to rewrite the SIP URI? Currently I have the following [EMAIL PROTECTED] I am setting it so it does the change but its still showing up with the prefix. I need it to look like this: [EMAIL PROTECTED] I got xxx.xxx.xxx.xxx to change to yyy.yyy.yyy.yyy I just need the prefix to go away now.. ☺ ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Little OT.. SER Question
Thanks for the info here is a sniplet of how I am doing my change now
with a php script calling to a MySQL DB. Where would I insert the
strip(5)
command?
# resolve alphanumeric names in the URI
if (uri=~sip:[EMAIL PROTECTED]) {
log(Running astSer.php \n);
exec_dset( /usr/local/etc/ser/astSer.php );
};
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Colp - Asterlink
Sent: Thursday, November 10, 2005 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Little OT.. SER Question
And so the file said to the Brian... Let there be enlightenment:
strip(5);
That'll strip off the first 5... Characters... From the URI
Joshua Colp
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RE: [Asterisk-Users] Uninstall AMP
rm rf / ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uninstall AMP Hi! How do I uninstall AMP and FOP from my Asterisk? Regards Anders Svensson This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incomming calls
the i is if you were to press an incorrect digit. s is for START. You can also specify your DID as a start point. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pablo Allietti Sent: Tuesday, November 01, 2005 10:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Incomming calls Hi all i have a question. is my first time using [EMAIL PROTECTED] and i need your help i configure all my asterisk to go outside and work perfect via te110p but now i need to receive calls. but when in my PBX i digit the number for example 202 the asterisk receive a s i suppouse. the error message is that -- Going to extension s|1 because of Complete received -- Executing Playback(Zap/31-1, vm-goodbye) in new stack -- Accepting call from '' to 's' on channel 0/31, span 1 == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' can anybody help me with the instructions ? -- .- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1.2beta2 and spandsp
Can I get a copy of your makefile? I am having a devil of a time getting it to work.. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Sent: Tuesday, November 01, 2005 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.2beta2 and spandsp Nothing special... yes I patched them manually -- current spandsp patch is broken a little. On Tuesday 01 November 2005 18:11, Anton Krall wrote: Any special considerations? How did you patch the files? Manually? | |On Tuesday 01 November 2005 12:19, Anton Krall wrote: | Anybody already tried compiling spandsp with the new 1.2beta2? | How about unicall? | |I did, and it works fine. |What is unicall? |___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_txfax.so app_rxfax.so
ok.. followed instructions with the apps but I keep getting this error: [app_rxfax.so]Oct 30 21:08:48 WARNING[15290]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handlerand [app_txfax.so]Oct 30 21:09:16 WARNING[15317]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info I am using the download of 0.0.2PRE21a along with head from Friday. I have been told using 0.0.3 will cause this but I have never downloaded or installed it. Only version 0.0.2. Any ideas? [EMAIL PROTECTED] The world at your fingertips... Brian C. FertigData Telecom Engineer Planet Telecom, Inc.4701 W. Hillsborough AveTampa, FL 33614 tel: fax: mobile: SiP URi: 813-864-3161813-881-9762813-817-9961[EMAIL PROTECTED] This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!
How do I resolve this? -- Unregistered SIP '107' -- Registered SIP '107' at 192.168.0.161 port 5060 expires 60 -- Registered SIP '107*' at 192.168.0.161 port 5066 expires 60 Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' This happens and all of my phones loose registration. Its driving me nuts. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- AIM: planetTelNOC ICQ: 65075522 MSN: [EMAIL PROTECTED] This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: T1 questions follow-up
Sorry I felt left out.. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michaël Gaudette Sent: Thursday, October 20, 2005 2:37 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: T1 questions follow-up Tom, Thank you! This was all hypothetical, because I'm trying to wrap my mind around the concept. But you've made it much clearer for me. I still have a few follow-up questions... 1a) Forget the hypothetical company now. Let's say 6 outside lines were deemed sufficient, and there were 12 employee (i.e. inside lines). Could I have the same Digium T1 card to service out and inside the company? Yes. You can order a T1 ISDN PRI from most carriers and they can limit the circuit amount on the T1. 1b) I'm fairly certain of this, but anything going outside, I could use the same T1 for receiving calls as for sending them, right? (not with the same channel at the same time, obviously, but I could use 2 lines incoming and 4 outgoing, or 3 incoming and 3 outgoing, depending on the current situation, without reconfigurin the PBX?) Yes you have to tell them that you want a Bi-Directional PRI. Yes. You would use the group feature for Zaptel. The call would be something like: dial(ZAP/g1) this would take the next channel available and send it out. 2) When you say a PRI is necessary for a Caller-ID name (as opposed to just number), I've looked around and I understand a PRI uses a T1 tunnel (24 channels) but a bit more expensive. Is this a fact, or is it something completely separate that I couldn't use with Asterisk? Or am I completely out in left field? What you want to order is ISDN PRI NI signaling. CAS will not cut it for you. Yes. Normally you would have 24 channels in a T1. However they can fraction it off for you with 1 D channel and X amount of B channels. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call to all Astricon attendee's!!!!
If you were at Astricon 2004, Astricon Madrid, Astricon 2005 or ANY other astricon event and you have pictures we would love to have them. Please drop us a email and we will make arrangements to get your pictures from you. We are located at: http://astri2005.netdr.biz Brian Fertig This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: www.openpbx.org
Can they do this? Is this legal? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith Sent: Friday, October 07, 2005 10:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: www.openpbx.org Further info. The domain is registered to Marc Olivier Chouinard. He has posted in the dev list. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: www.openpbx.org
sigh.. meaning take the fork ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Friday, October 07, 2005 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: www.openpbx.org Brian C. Fertig [EMAIL PROTECTED] wrote: Further info. The domain is registered to Marc Olivier Chouinard. He has posted in the dev list. Can they do this? Is this legal? Yes - anyone can register a domain name. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DPH-140S SIP Phone oddities
Title: RE: [Asterisk-Users] DPH-140S SIP Phone oddities Change your DTMF setting to rfc2833. You may be using an incompatible type with your phones. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Juan Janczuk Sent: Tuesday, October 04, 2005 12:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] DPH-140S SIP Phone oddities Hi, list! I'm playing on an [EMAIL PROTECTED] installation, since a month or two. I've had no trouble setting it up 'n running. I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk. From this phones, I can make receive calls with no trouble, but, when I try to use some interactive function (eg Directory or Voicemail), the phone seems unable to transmit the digits to Asterisk. With the same config, but with a softphone (X-Lite), the digits are transmitted with no trouble at all. Please, do anyone have any clue? Thanks in advance. Juan. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.9/116 - Release Date: 30/09/2005 File: ATT207437.txt This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections
are you giving answer()? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=3 udpStart=20001 udpEnd=3 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c21.15.3-4 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN-GATEWAY
Well the simplest is to make the connection insecure with a static ip. sip.conf [cisco2600] host=xxx.xxx.xxx.xxx defaultip=xxx.xxx.xxx.xxx insecure=yes type=friend disallow=all allow= (your codecs) extensions.conf [default] ;dial out cisco exten = _1X.,1,Dial([EMAIL PROTECTED]) As far as your cisco config for voice, depends on hardware and how you want to setup your dialpeers. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office 813.864.3161x107 Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reli Loin Sent: Wednesday, September 28, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PSTN-GATEWAY Hello, I have just installed asterisk and I would like to connect it to the PSTN. I have a gateway Cisco 2600, how must I declare it in the file of configuration (extensions.conf, sip.conf). thanks for your helping ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN-GATEWAY
Well the simplest is to make the connection insecure with a static ip. sip.conf [cisco2600] host=xxx.xxx.xxx.xxx defaultip=xxx.xxx.xxx.xxx insecure=yes type=friend disallow=all allow= (your codecs) extensions.conf [default] ;dial out cisco exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED]) Sorry forgot the SIP/ part. :) As far as your cisco config for voice, depends on hardware and how you want to setup your dialpeers. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office 813.864.3161x107 Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reli Loin Sent: Wednesday, September 28, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PSTN-GATEWAY Hello, I have just installed asterisk and I would like to connect it to the PSTN. I have a gateway Cisco 2600, how must I declare it in the file of configuration (extensions.conf, sip.conf). thanks for your helping ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323
yes.. I have looked. they are different. But when I unregister 1 the other will register.. Its only when I have 2 of them trying to register at the same time I have an issue. But yes the ID's are different in both of them. b From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Fri 9/23/2005 2:15 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323 On 9/23/2005, Brian C. Fertig [EMAIL PROTECTED] wrote: I am having a slight issue. I am trying to register 2 asterisk boxes with GNUGK and when I try to add the 2nd it gets denied cause of it saying its a duplicate. How do I change the configs to allow more than one asterisk box register to the same GK? brian Don't 'quote me' on this but... Look in the h323.conf/s and see if you have two different h323id strings for the servers. I think it defaults to 'Asterisk PBX' but the current sample makes it 'ObjSysAsterisk'. I am pretty sure they have to have different names or GNUGK is going to think they are the same. Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323
I am having a slight issue. I am trying to register 2 asterisk boxes with GNUGK and when I try to add the 2nd it gets denied cause of it saying its a duplicate. How do I change the configs to allow more than one asterisk box register to the same GK? brian This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AstriCon 2006 Location
Im thinking Tampa or Orlando Florida! Nice warm.. Granted you may have to dodge hurricanes.. But hay its worth it! ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Younger Sent: Monday, September 19, 2005 10:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] AstriCon 2006 Location Great Idea! I suggest Sydney :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Tuesday, 20 September 2005 12:02 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] AstriCon 2006 Location What have it outside the US j/k -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni Miano Sent: Monday, September 19, 2005 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AstriCon 2006 Location The best place for Astri Con 2006 would definatly be Naples, ITALY __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] rotate * log file?
Try this after your done rotating your log: asterisk -rx reload This is what I use.. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Sunday, September 11, 2005 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] rotate * log file? Rich Adamson wrote: Running fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rv). Would like to rotate the files without killing the cli session. Any reasonable way to accomplish this? Rich man logrotate ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fedora Core 4 x86_64
Take it from someone who owns 25 of them. Stay away from FC anything. Use CentOS 4 its better more stable and has true multi-treading as FC doesn't thread anything.. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Supporter Sent: Friday, August 26, 2005 1:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Fedora Core 4 x86_64 I am about to build a Dual Opteron Asterisk box as our soon to be production server. Is Core 4 supported or should I stay with Core 3? There was a recent post about an issue with the latest Core 3 Kernel and zaptel. I had the same experience, but just rolled back to the previous version of the Kernel on Core 3 on our evaluation server. Thanks in advance ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which AGI Development Software is fastest onAsterisk?
What can you develop in? What are you comfortable? I use PHP for testing then convert into C shared objects. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Thursday, August 18, 2005 4:02 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which AGI Development Software is fastest onAsterisk? I'm looking to develop some custom AGI that will be MySQL intensive. It appears Asterisk supports many different development environments. Which would be best suited for Asterisk and MySQL? Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: DTMF, Asterisk, External PSTN gateway, and PAP2 (was: RE: [Asterisk-Users] Issue with DTMF Tones - CodecIssues)
I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, August 16, 2005 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: DTMF, Asterisk, External PSTN gateway,and PAP2 (was: RE: [Asterisk-Users] Issue with DTMF Tones - CodecIssues) This is not an answer but rather an addition to the question. We're using a large scale VOIP only asterisk system that has PAP2 enduser units using inband as their DTMF mode. sip.conf is set for using inband as well, and we pass PSTN calls through a provider. Here's the problem, when our users call other IVR systems like banks and other voicemails they have been unable to always pass the keys they press. Sometimes if they press the keys slowly it works, but not all the time, and otherwise it definitely doesn't work. Anyone else had this problem and/or know of a possible solution? Sherwood This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipjet experiment
I get the same problem @ home when I use it. I thought it was just me. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey Sent: Friday, August 12, 2005 10:58 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Voipjet experiment Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is causing the problem, or if VoipJet doesn't sense the line was picked up (and thus doesn't pass this info to me). Here is a sample output of CLI when the disconnect happens: --- -- Called voipjet/18004337300 -- Call accepted by 216.118.117.46 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-1 is making progress passing it to SIP/541-d994 -- Nobody picked up in 3 ms -- Hungup 'IAX2/voipjet-1' --- During the 3 ms I hear the American Airlines auto attendant giving me options, I can choose and option and the auto attendant will recognize the DTMF and send me to that menu, then after a total of 30 seconds, I get disconnected. I haven't had this issue with any other numbers yet (only been in production use one day...) Any info is appreciated. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with TNT and Asterisk
Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mahler's Book - New Project
Goto their website and buy it. www.signate.com I know paul he's a good guy. Has a new book coming out soon. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Stude Sent: Wednesday, July 20, 2005 9:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Mahler's Book - New Project Hi all, I'm currently gearing up for a possible PBX replacement project using Asterisk, and I'm just breaching the iceberg of information that's available. I typically like to have something thick with pages in front of me. Mahler's book was the first one to come up and it seems like a good place to start. However, the big name bookstores tell me it'll take up to three weeks, and this project simply can't endure that wait. Does anyone know where it's possible to get a paper copy *quickly*? #2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can anyone recommend a reference book or site more suited to this task? Thanks and regards, David Stude Receptec, LLC Holly, MI This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Comments on Areski Calling Card Solution plz
I have played with it. I don't know how well it would work for production. Maybe with some custom coding etc you could get it to do what you want. But out of the box its good for testing and nothing more. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arnd Vehling Sent: Monday, July 18, 2005 12:18 PM To: Asterisk Users Subject: [Asterisk-Users] Comments on Areski Calling Card Solution plz Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations
Trust me dude.. You don't want a lucent TNT. If your going all out for an DS3 and you don't want to multiplex it then you will need something that will take a DS3 which I don't believe TNT's do. Purchase an AS5400HPX they will and work very well. Set yourself up with some dialpeers etc and your good to go. Trust me. I have done it. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Wednesday, July 13, 2005 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations At 10:06 AM 7/13/2005, you wrote: Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM - VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features are needed, I think it can handle the routing. If not, I can whip up a SER box. We currently have a Cisco 7206VXR (1 voice resource) and a Cisco AS5300 (120 voice resources). The DS3 will also have SS7 signaling on it. Recommendations/comments/concerns/rants are graciously welcomed. Lucent TNT Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to dial certain calls
First off kill the Glaw. It doesn't exist.
Then try your call. But also why are you sending the line congestion
when you first start to make a call. That's normally used as a closure.
But from what I can see about the only thing wrong is the GLAW. Kill
that and you should be good to go.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JP Russell
Sent: Tuesday, July 12, 2005 5:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unable to dial certain calls
Of course. Note that I have no idea what glaw is but
someone on some board
shttps://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gif
https://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gifugge
sted
it as a resolution to a similiar problem so I put it in.
The entry from the iax.conf file is:
[vbx]
type=peer
host= 213.61.187.150
secret=-my password-
notransfer=yes
context=def
allow=glaw
allow=ulaw
allow=gsm
and from extensions.conf I guess you need the [def]
context entries.
they are:
;NL
exten = _00316.,1,Congestion
exten = _00319.,1,Congestion
exten = _0031X.,1,SetCallerID(Not Available
7005551212)
exten = _0031X.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _0031X.,3,Playback(invalid)
exten = _0031X.,4,Hangup
;US
exten = _001X.,1,SetCallerID(Not Available
7005551212)
exten = _001X.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _001X.,3,Playback(invalid)
exten = _001X.,4,Hangup
Finally sip.conf includes the below paramaters:
[general]
disallow=all
allow=ulaw
allow=glaw
allow=gsm
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = from-sip ; Default for incoming calls
callerid=No CallID
[2203]
port=5061
username=-thisusername-
secret=-this password-
host=dynamic
type=friend
nat=1
qualify=no
;reinvite=no
canreinvite=yes
context=intern
On Mon, 11 Jul 2005 22:55:49 -0400
Brian C. Fertig [EMAIL PROTECTED] wrote:
Check your codecs.. Can you post a sniplet of your IAX,
SIP, and extensions.conf for dialing the US so we can see
were the problem may lie?
Brian Fertig
From: [EMAIL PROTECTED] on behalf
of JP Russell
Sent: Mon 7/11/2005 9:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to dial certain calls
To begin with, I am new to both asterisk and VOIP and
although I've
gotten pretty far with my Asterisk setup and have two
different sip
accounts working fine for outgoing calls I am having
trouble with one
issue.
My problem is that I have another provider who uses IAX2
that I wish
to use for calling various countries, including local
(The
Netherlands) calls and calls to the US to name two. I
am able to
call local numbers without a problem through this
provider with
Asterisk, but calling US numbers is not working.
I CAN call the same US numbers with the service by using
a direct
connection from a softphone for example.
The entries that show up in the log after failed
attempts to call the
US are:
Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line
1851
(ast_channel_make_compatible): No path to translate from
SIP/2203-2929
(4) to IAX2[vbx]/1(16)
Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line
672
(dial_exec): Had to drop call because I couldn't make
SIP/2203-2929
compatible with IAX2[vbx]/1
I don't see anything suspicious entries in the CLI
logging with IAX2
debugging on. Searching the archives and google didn't
turn up a
solution to this or even point me in the right direction
I'm afraid.
Anyone have any idea on what my problem is or I can go
for this issue?
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RE: [Asterisk-Users] asterisk and h.323
To answer your question is this a router? I am not aware of this model being able to do voip. I am fluent in Cisco VOIP configs but I dont know this one. I just did some checking and this router will not do voip as far as I can tell. I believe the smallest model is a 2600 series that will do voip but your TDM voice card will cost you. they arent cheap even used. From: [EMAIL PROTECTED] on behalf of Todd Reese Sent: Mon 7/11/2005 5:42 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk and h.323 Hi All, I just purchaced a Cisco uBR924 and was under the assumption that it did SIP. Being somewhat new to Asterisk, is there anyone willing to supply a working config that will get me started on configuring these items. Best Regards winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to dial certain calls
Check your codecs.. Can you post a sniplet of your IAX, SIP, and extensions.conf for dialing the US so we can see were the problem may lie? Brian Fertig From: [EMAIL PROTECTED] on behalf of JP Russell Sent: Mon 7/11/2005 9:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unable to dial certain calls To begin with, I am new to both asterisk and VOIP and although I've gotten pretty far with my Asterisk setup and have two different sip accounts working fine for outgoing calls I am having trouble with one issue. My problem is that I have another provider who uses IAX2 that I wish to use for calling various countries, including local (The Netherlands) calls and calls to the US to name two. I am able to call local numbers without a problem through this provider with Asterisk, but calling US numbers is not working. I CAN call the same US numbers with the service by using a direct connection from a softphone for example. The entries that show up in the log after failed attempts to call the US are: Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line 1851 (ast_channel_make_compatible): No path to translate from SIP/2203-2929 (4) to IAX2[vbx]/1(16) Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line 672 (dial_exec): Had to drop call because I couldn't make SIP/2203-2929 compatible with IAX2[vbx]/1 I don't see anything suspicious entries in the CLI logging with IAX2 debugging on. Searching the archives and google didn't turn up a solution to this or even point me in the right direction I'm afraid. Anyone have any idea on what my problem is or I can go for this issue? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI or Trunk monitoring
3650 what? Cisco doesn't make a 3650.. ..o---o. Brian Fertig NOC/Network Engineer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Tuesday, July 05, 2005 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI or Trunk monitoring Like many others (I suspect) I have a Sipura 2000 attached to my fax machine which is in turn connected to my * box which in turn has a T1 terminated directly into it. This setup has proved to be 100% reliable for the 6 months (or so) that it's been in service. I'm running a 100Mb switched network based on the Cisco 3650. Mark Carlos Alperin wrote: Did someone monitor the PRI's or trunks some way? I tried with MRTG and Andrea Fino module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Visual ring notification
How good is your electrical engineering? ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Friday, July 01, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Visual ring notification I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Got SIP response 481 Invalid CSeq Number backfrom X.X.X.X
I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Alves Sent: Friday, July 01, 2005 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got SIP response 481 Invalid CSeq Number backfrom X.X.X.X I get this error message when sending calls to a Cisco Gateway AS 5300, one call out of 10. Is there any configuration hack either on Asterisk or the Cisco that would this problem go away?? Federico Alves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Got SIP response 481 Invalid CSeq Number backfrom
as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Alves Sent: Friday, July 01, 2005 5:15 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got SIP response 481 Invalid CSeq Number backfrom I am using G711. In the Cisco, how many bytes should I use for the payload? Is there any way to configure the payload in Asterisk? Thanks in advance. Federico Alves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Voip Question
Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need a little in sight on how to setup the dial peer or something in the global config for the router. TIA ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Logrotate
I created some scripts to logrotate. I am having a problem. After I do it, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Logrotate
Asterisk doesn't use the syslog daemon tho does it? I thought it did internal logging to a file. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geoff Manning Sent: Thursday, June 30, 2005 10:30 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Logrotate Could someone help me out with how I can rotate asterisk's log's without killing the process? Does restarting the syslog service help? # service syslog restart or # /etc/init.d/syslog restart This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Logrotate
thank you I will give that a try.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hilton
Williams
Sent: Thursday, June 30, 2005 10:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Logrotate
Hi
[EMAIL PROTECTED] uses the following file:
/var/log/asterisk/*log {
missingok
rotate 5
weekly
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
/var/log/asterisk/full {
missingok
rotate 5
daily
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
/var/log/asterisk/cdr-csv/*csv {
missingok
rotate 5
monthly
create 0640 asterisk asterisk
}
It connects to the running asterisk and issues the command logger
reload.
Regards
Hilton
Datatex Dynamics CC
Web site http://www.datatex.co.za/
Email to [EMAIL PROTECTED]
Tel +27215924033
Fax +27215924077
The use of the Datatex e-mail facility is not permitted
for the distribution of chain letters or offensive email
of any nature whatsoever. Datatex hereby distances itself
from and accepts no liability in respect of the
unauthorised use of its e-mail facility or the sending of
e-mail communications for other than strictly business
purposes. Datatex furthermore disclaims liability for any
unauthorised instruction for which permission was not
granted. Any recipient of an unacceptable communication,
a chain letter or offensive material of any nature is
requested to report it to [EMAIL PROTECTED]
- Original Message -
From: Brian C. Fertig
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, June 30, 2005 4:22 PM
Subject: [Asterisk-Users] Logrotate
I created some scripts to logrotate. I am having a problem. After I do
it, I am sending kill -HUP to the process
its not using the newly created messages file again. Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
This email was scanned by: Mcafee GroupShield
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Use of this information by anyone other than the recipient or
sender will be considered in breach of agreement.
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RE: [Asterisk-Users] VOIP-INFO down?
I will host a mirror also before long. I am moving to a new DC and will have more bandwidth available. Brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, 14 June, 2005 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VOIP-INFO down? I also would be willing to host it.. or load balance it...and I also run a regional ISP in the northeast (I'm just below John state wise hehehe) On 6/14/05, John Bittner [EMAIL PROTECTED] wrote: To whoever owns this site. To help keep this up and running I am willing to host it for free. I run a regional ISP in the northeast. Please contact me off list. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 14, 2005 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] VOIP-INFO down? Second day in a row... -Original Message-MYDYNDNS.ORG From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi all, Is VOIP-info down? Marcel van Kaam Fonetica Teleservices ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation
Just use a cisco with 5 T1 ports and have everything over IP use ultra monkey to load balance your asterisk boxes. I have found this works very well. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Monday, 13 June, 2005 11:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Variables and status problems in AGI application
I am having the same problem. I have opened a bug report on Digium's website about it. I found it stopped working sometime at the end of april and would like to roll back to that version. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 06 June, 2005 11:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Variables and status problems in AGI application I am running a prepaid application with Asterisk. When authentication has to be done by DTMF everything works fine. However when the user is authenticated directly from the sip phone, the channel variables seems to disappear. Trying to retrieve the channel status always returns -1 instead of the 6 that happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME variables. The only difference in the program is that in the DTMF mode, commands are sent to to the AGI channel. Is there a reason why? PS: The situation has now reversed. SIP calls retain the proper status and PSTN calls lose their status value, always returning -1. This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHPAGI problems
First off.. Just do a: exten = 12345,1,AGI(dtmf) And try running your php from the console and see if you get debug issues. .o---o. Brian Fertig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hariharan Gopalan Sent: Tuesday, 24 May, 2005 16:54 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PHPAGI problems Hi Here is part of my extensions.conf exten = 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (gsm|ilbc|speex), priority = mine -- Executing AGI(IAX2/[EMAIL PROTECTED]:4569-2, dtmf.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dtmf.php -- AGI Script dtmf.php completed, returning 0 -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-2' When I dial, this does not do anything, and just disconnects. But if I dont use the phpagi.php, I am able to run a simple php agi script well. Wonder what I am doing wrong. Any help is highly appreciated Thanks Hariom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip to sip
If your looking to link 2 asterisk boxes might I suggest IAX. Much more efficient in the way bandwidth is utilized between the locations. Also if you want to use your sip solution, have you setup the other end point in your SIP.CONF? I have never got IP dialing to work in asterisk but it works fine when assigned in the conf file. .o---o. Brian Fertig NOC/Network Engineer Systems Engineer From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Quintin Sent: Monday, 23 May, 2005 08:08 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sip to sip Hi Im trying to put up an sip pbx system for my company but im getting some problems when Im trying to call from server ( branch A ) to server ( branch B ) This is my extentions.conf : exten = 3003,1,Dial,SIP/[EMAIL PROTECTED] And this is what I get when I try to dial that user in branch B _ -- Executing Dial(SIP/5001-66b1, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 404 Not Found back from 192.168.0.200 -- SIP/192.168.0.200-e638 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/5001-66b1' status is 'CONGESTION' Both servers are exactly the same.. What can the problem be, that branch B server doesnt route the call through Thx Quintin This email was scanned by: Mcafee GroupShield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip to sip
Well yes and no. If they have static IPs then you only need to setup a context as such: You would assign the following information on your Branch B server with BranchAs information. [branchA] type=friend defaultip=xxx.xxx.xxx.xxx context=default insecure=yes host=xxx.xxx.xxx.xxx disallow=all allow=g729 allow=alaw allow=ulaw You would do the same here but for the Branch A server with Branch Bs config. [branchB] type=friend defaultip=xxx.xxx.xxx.xxx context=default insecure=yes host=xxx.xxx.xxx.xxx disallow=all allow=g729 allow=alaw allow=ulaw In your extensions.conf your dialplan would look something like this: exten = _30.,1,Dial([EMAIL PROTECTED],23,r) ; use this for calling people on branch B There is no need to register the boxes with each other if they are static, which is the easiest way to set this up. Any other questions lemme know.. .o---o. Brian Fertig From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Quintin Sent: Monday, 23 May, 2005 09:23 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sip to sip Hi B Do you mean I must do this in my sip.conf file on eatch server Branch A register= 3001:[EMAIL PROTECTED] /3001 Branch B register= 5001:[EMAIL PROTECTED] /5001 thx Q From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Brian C. Fertig Sent: 23 May 2005 03:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] sip to sip If your looking to link 2 asterisk boxes might I suggest IAX. Much more efficient in the way bandwidth is utilized between the locations. Also if you want to use your sip solution, have you setup the other end point in your SIP.CONF? I have never got IP dialing to work in asterisk but it works fine when assigned in the conf file. .o---o. Brian Fertig NOC/Network Engineer Systems Engineer From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Quintin Sent: Monday, 23 May, 2005 08:08 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sip to sip Hi Im trying to put up an sip pbx system for my company but im getting some problems when Im trying to call from server ( branch A ) to server ( branch B ) This is my extentions.conf : exten = 3003,1,Dial,SIP/[EMAIL PROTECTED] And this is what I get when I try to dial that user in branch B _ -- Executing Dial(SIP/5001-66b1, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 404 Not Found back from 192.168.0.200 -- SIP/192.168.0.200-e638 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/5001-66b1' status is 'CONGESTION' Both servers are exactly the same.. What can the problem be, that branch B server doesnt route the call through Thx Quintin This email was scanned by: Mcafee GroupShield This email was scanned by: Mcafee GroupShield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] play gsm files in windows
Eric, Do you know of one that can convert or record? .o---o. Brian Fertig NOC/Network Engineer Systems Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Monday, 23 May, 2005 11:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play gsm files in windows Use Apple QuickTime Best Regards Erick W. - Original Message - From: Brett, Gary [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 23, 2005 9:34 AM Subject: [Asterisk-Users] play gsm files in windows Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programs to parse queue_log
I am in the process of doing mine now. It works ok here and there not 100% as of yet. But its written in PHP .o---o. Brian Fertig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johann Sent: Monday, 23 May, 2005 11:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Programs to parse queue_log What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
That in now way shape or form was funny. I about had a heart attack when I was reading this. To move to a winDOZ platform would just make asterisk SUCK! But its nice to know its staying where it is. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Friday, 01 April, 2005 12:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now Remco Barende wrote: On Fri, 1 Apr 2005, Chris Hills wrote: Olle E. Johansson wrote: * New source code structure - C# and .net Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. Therefore, the code was quickly moved to C# on the .net platform. This gives Asterisk a lot of new features, including being fully integrated with Microsoft Exchange and Microsoft Active Directory. With all the user data stored in Active Directory, we finally have the user under full control. Users can dial in to the PBX to change their Windows password. We can also implement single-sign-on based on DTMF from a cell phone or WiFi phone. says Kelvin Reming. The C# language gives us much more modern code. And I'm so happy to get rid of the stupid-looking arctic bird, an ugly animal that that couldn't even fly. Shame this is just an april fool, I like the sound of this! Though it would be going head to head with Live Communications Server... I guess you missed the real joke there (the stability and secureness of .net) Ya, I mean do you really think an open source community is gonna acknowledge that MS can do anything right? of course not. THEY'RE THE DEVIL! (note, I will not respond to anything posted in reply to this, so don't even try) -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HELP: Dose G.729 with IPP only worked on Intel CPU?
yes it only works on INTEL. Good luck otherwise. I have tried.. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Friday, 18 March, 2005 12:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HELP: Dose G.729 with IPP only worked on Intel CPU? Hi, ALL: I install IPP(l_ipp_ia32_itanium_p_4_1_2.tar) and download the speech codeing (l_ipp-sample-speech-coding_p_4.1.008.tgz) then patch it (g729-041103.diff). My CPU is Centaur VIA Nehemiah with 998.715 MHz processor not INTEL CPU. I choose PIII as its CPU type when I modify Makefile under G729-float. # For PIII OPTIMIZE= -O6 -mcpu=pentium3 -march=pentium3 -ffast-math -fomit-frame-pointer IPPCORE=a6 I got the codec_g729.so and copy it to /usr/lib/asterisk/modules/. Modified /etc/init.d/asterisk and add LD_LIBRARY_PATH and export it. Modified /etc/asterisk/sip.conf and add allow=g729. I worked my asterisk well before add G.729 codec. But after it, my asterisk crashed a few seconds after I run a startup command /etc/init.d/asterisk start. Does anyone have the same problem? -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec negociation
If you don't want to proxy the media through * the put this setting: canreinvite=yes this will allow the 2 end points to connect directly for the RTP bypassing you. otherwise I have noticed the same when I try to proxy I have to make sure everyone is using the same codec or it doesn't work well. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yves Sent: Thursday, 17 March, 2005 12:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Codec negociation Hi, I've got an Asterisk latest CVS head with oh323 installed. There is one thing I can't understand about the codec negociation. I receive calls in G723G729, and send them to another gateway who can handle both codecs too. So all I want to do is just passthrou, for both. It seems that * only try to send with the first of the list, what is fine when it's the good one, but otherwise he complain about being unable to transcode instead of trying the second codec. I hope I've explained well my problem. Could someone explain me a little bit more about the negociation ? Or did someone have the same issue ? I didn't find much info, tried docs google. Thank you. Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Errors
Can someone explain what this error is? -- Got SIP response 500 Server Internal Error - Invalid CSEQ number back from 209.xxx.xxx.xxx How do I fix this? .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] APP_QUEUE MYSQL LOGGING
Does anyone know if this has been implemented? I have been around the sites and haven't really found much. I know there was an old patch that would make it work but it doesn't do anything but break the application now. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hung Sip Channels
Does anyone know how to get rid of these hung channels? I am getting this when I do a: show sip channels 209.82.xxx.xxx0071495217 2591218534@ 00103/1 unknow(d) 209.82.xxx.xxx0041590104 0690231739@ 00103/1 unknow(d) 209.82.xxx.xxx0070259259 3265102826@ 00103/1 unknow(d) 209.82.xxx.xxx0071948143 1927207026@ 00103/1 unknow(d) 209.82.xxx.xxx0022576786 1752809624@ 00103/1 unknow(d) 209.82.xxx.xxx0070153955 0085223171@ 00103/1 unknow(d) I have about 60 of them and growing. I have submitted a ticket with my provider to let them know of this problem but I would like to clear them out w/o restarting the asterisk binary. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys RT31P2-NA
I am noticing a problem with the RT31P2-NA when it loses internet. Has anyone experienced problems where it does not reconnect to asterisk and obtain its dialtone again? Brian Fertig Planet Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is it possible to ID payphone calls?
Yes it is possible via the ISDN OLI. It will tell you what the call is originating from. Not sure if * will decode the OLI or not. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jess Coburn Sent: Monday, January 17, 2005 11:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is it possible to ID payphone calls? Hello I have a 800 DID setup to dial into my Asterisk server and I'm wondering if it's possible to ID when it's a payphone or not? I suspect it's not since I'm getting calls from someone else's SIP or IAX box. If I had a digium card installed and connected to a couple lines would I be able to get this information and parse it? Thanks, Jess ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec conversion
In your SIP.CONF you need to tell * what codecs to use. sip.conf [broadvoice] disallow=all allow=ulaw [phone] disallow=all allow=g729 Then in your extensions.conf you just have it dial as usual. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Helder Rogério [MICROREDE] Sent: Monday, January 17, 2005 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Codec conversion Hi! Is there any way to receive in * server a call from a Terminal adapter in G.723/G.729 and then convert it to G.711? I'm wondering this because I can only place all thru Broadvoice in G.711 but most of customers have ADSL connection with 128k upstream, so the result is that they can hear in excellent conditions but can't be heard very well the sound is all choppy. even directly to broadvoice thru Xten sip client. So the idea was to act as proxy and codec converter so that the communication coming out their router is the smaller it can get. I've mentioned G729 or G.723 becuase their routers have it, (Draytek 2600V). Thanks in advance for your suggestions Helder Rogerio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS Gateway
Does anyone know of any companies where I can interconnect with for SMS? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 and Asterisk solution
If you are looking for a SS7 solution right now with out paying anything more for asterisk you can purchase a solution from Verisign called SIP-7. You send your signaling to them and they send the RTP to your media gateway. From what I understand its very efficient and offers all the same features as SS7 does. They also have an interconnect with Cable and Wireless in the UK for services there. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Wednesday, January 12, 2005 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SS7 and Asterisk solution On Wed, Jan 12, 2005 at 05:30:31PM -, Ben Merrills wrote: We have the problem that our telecoms provider deals mainly in SS7 (C7, and it seems most in the UK do). For us to take EuroISDN off them, with the same features as SS7, we have to be put through a protocol converter, now this isn't an issue for us, but it is for them. Most UK phone companies (i.e. BT or the smaller regional carriers) all use SS7, everywhere! For the most part they don't accept VoIP termination (although I think BT might have some facilities for this). So they very much try and push SS7 on interconnects. And that's why SS7, for me (and I think for quite a few others taking PRI style links in the UK) is so important. Unfortunately SS7 comes at a cost. In the UK to talk to a telco using SS7 you generally needed a Telecomms license (which mandates telcos to interchange traffic with you). Now telco licenses have been scrapped (as per EU directives and the Communications Act) you're just meant to be able to ask etc. However they can demand that your SS7 stack is certified, and BT take about 6 months to provision/test an SS7 voice interconnect, other telcos may take longer. If you scr*w up at the SS7 level, they'll disconnect you as fast as you can shout sorry, and they can refuse to interconnect with you ever again !!! You also have to be running the right version of SS7 (I think the latest is UK8, though a lot of operators are running UK7). Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
I agree.. No certs needed. I know * better than probably all of your students combined dude.. I agree with BKW.. .o---o. Brian Fertig Network Engineer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 22, 2004 10:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Another Asterisk Certification No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hung SIP channels in Asterisk
Can someone tell me how to clear hung SIP channels in asterisk without restarting? Currently I have 62 channels and only show 10 in use.. this is some of the sip show channels output.. xxx.xxx.xxx.xxx00xxx24xxx 04240xx 00103/1 UNKN (d) How can I remove these? from * without rebooting? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail.Conf
When I specify the users voicemail can I specify more than one email address to send the recording to once its finished? Also can I set it where it only emails the voicemail recording and not stores it local to the * box? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping out of Queue to voicemail
When I setup Queuing I wasn't to give the user the ability to drop out and leave a voicemail. ok to accomplish this I understand I have to set the context in the queues.conf file. Now I have done this but when I go to invoke the voicemail function so they don't have to wait in queue it doesn't work. It only seems to work when it tried to dial one of the agents. Can someone give me some pointers on how to accomplish this and streamline it a little more? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Total newbie here looking to do a VoIPconference call?
Thanks for that. I just got rid of packet 8 and went with 100% asterisk in my house. But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But would like to have an extra FXS laying around just in case.. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Friday, December 17, 2004 5:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Total newbie here looking to do a VoIPconference call? My packet8 dta310 adapter has the SIP server hardcoded into it. If I could change that, I could use that? Search on broadbandreports.com VoIP forum - there are several postings (including a few by me) with instructions on how to downgrade the DTA-310 to v, put in the SIP settings and upgrade to (not higher than) v1234. I haven't tried it with *, but I assume it should work. -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeelatjafferali.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring a call in an Call Center Environment
How can I monitor calls in a call center environment real time? Is this possible? If so could someone show and example of how this is accomplished? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue question
Is it possible to have customers to be in queue and have a prompt that asks them if they want to leave a phone number so when there time is up they will get a call back so they can speak with the CSR? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queue monitor
You can setup recording by default. This is how I have mine setup. I don't believe the way app_queue is now you can have the agent press something to have it start recording. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruben Santos Sent: Wednesday, December 01, 2004 4:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] queue monitor We want to set up monitoring of calls going into our queue. We want to know if there was a way to initiate it, by having the agent who picks up the call dial a number to initiate the recording. Ruben T. Santos Director of Network Operations Brand X Networks (866) 487-3244 x 5203 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_queue question
But now in this instance it drops them into voice mail. Is there a way to have them punch in there phone number so they can keep there space in the system? Like if they are #20 in queue when they left their # for call back that when they get to number 4 or 5 that they would be called back and put back in queue to wait the remainder of time? Could this by chance be done with a AGI of some sort? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, December 01, 2004 4:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] app_queue question Check out queues.conf ; ; A context may be specified, in which if the user types a SINGLE ; digit extension while they are in the queue, they will be taken out ; of the queue and sent to that extension in this context. ; context = cytel-queuewaitnomore In our case, if a person presses 0 then they go into a generic voicemail box. Be sure to record a new message that lets them know of this. -Matthew - Original Message - From: Brian C. Fertig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 01, 2004 3:08 PM Subject: [Asterisk-Users] app_queue question Is it possible to have customers to be in queue and have a prompt that asks them if they want to leave a phone number so when there time is up they will get a call back so they can speak with the CSR? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Patch - estimated hold time announcements
Let me know if you find something out. I am having the same problem. I can get it to play to my agents but not the people on hold. I was debating on creating a AGI script to do all this but I remembered that it was supposed to do it automatically. If someone has a work around could you please share it? Brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Brussels Sent: Sunday, November 21, 2004 11:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue Patch - estimated hold time announcements On a pre 1.0 version I was running I patched queue.c to add estimated hold time announcements. Stable 1.0 (cvs checkout -rv1-0_stable asterisk) does not appear to of included this patch and of course patching the current queue.c with the patch I have fails. I looked at the Matis to see if an updated queue.c file is available or a patch for the current but I did not see anything - not too sure how these updates are included. Tried to pull the queue.c files w/ it's header files out of 1.02 but it would not properly compile. I also tried to compile the original queue.c file that was patched but it appears some header files it uses have been modified. Am I missing something here? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip.conf not paying attention to allow/disallow
I have found this same problem to be true. I don't know what to do to fix it. I believe it's a bug but don't know for sure. If you find a way drop me a line I would like to know. Brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, November 22, 2004 4:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip.conf not paying attention to allow/disallow In my sip.conf, under general I have: disallow=all allow=g729 allow=alaw allow=ulaw Then I have a specific sip: [RNK] clip disallow=all allow=alaw allow=ulaw allow=gsm If I do this: exten = _9.,1,Dial([EMAIL PROTECTED],60) The call still goes out as G729 even though I've told the RNK to disallow g729. I need to be able to make other 729 calls but to this one paticular group, they need to be 711. Any ideas? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] PoE switch question (Netgear FSM7326P works
I have a 3348 they don't do PoE. They do QoS and do it well. I don't know about the upper models.. brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, November 18, 2004 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] PoE switch question (Netgear FSM7326P works Eric Wieling wrote: Dell has some 48 port supposedly PoE switches for about $600. I've not done QoS on them, but they claim to support it. I don't see any PoE-enabled switches on Dell's web site, and the switch you are referring to (PowerConnect 3348) definitely doesn't have any references to 802.3af at all. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Controlling Asterisk from PHP?
No nothing exists. However may I suggest PHPAGI it's a class for asterisk to interface with it. You can pull channel variables etc and do all kinds of kewl junk with it. I write all my AGI in php and execute it. But yes you in a way can control asterisk with php at the AGI level. brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Vogel Sent: Thursday, November 18, 2004 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Controlling Asterisk from PHP? Hi! Is it possible to control Asterisk with PHP? I don't think that the extensions.conf can solve all my problems. So I would like to make it with PHP (which I really know well). I would need a possibility to read the dialed digits and a possibility to start a call. Does something like this exist? Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speaking of DS3s....
My thoughts are to have it demux'd on your end. break it into smaller T1's and bring them in that way. Your looking at like 2-3 PRI's per box depending on your config. This is the easiest way I could think of getting this to happen. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.881.9762 Fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, November 18, 2004 4:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Speaking of DS3s Scenario A: Lets say you had 10 Asterisk boxes, all 4U, 4 proc servers, all with same *.conf, in a rack mount unit. You can get 1 OC3 connection for $5,000 a month. How can you split that OC3 among the 10 boxes and have load balancing and auto-failover? Scenario B: Same setup as A, but this time, each of the 10 servers services 1 paticular area code (NPA). Lets say server #4 stops working. How do I auto-route an incomming call destined for #4 to another machine? (I'm guessing dundi here but other options welcome) Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Controlling Asterisk from PHP?
I am in the process of writing a book on the AGI structure of * but for now there are a couple examples on that site of how to implament it. I learned whatI know from voip-info.org most if not everything is there for what you may need to know. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Vogel Sent: Thursday, November 18, 2004 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Controlling Asterisk from PHP? Brian C. Fertig schrieb: No nothing exists. However may I suggest PHPAGI it's a class for asterisk to interface with it. So something exists ;-) Nope.. You have to create it. No COTS packages.. You can pull channel variables etc and do all kinds of kewl junk with it. I write all my AGI in php and execute it. But yes you in a way can control asterisk with php at the AGI level. Sounds cool. I just downloaded it. Now I only need to know, how to include it in asterisk. The documention is ... hmm ... ;-) Thanks! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reject a call if no callerID
But now that logic works. However how would you insert that into the
dialplan to get it to work or would AGI be better solution?
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Wednesday, November 03, 2004 3:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reject a call if no callerID
On Wed, 2004-11-03 at 18:45, Hermann Wecke wrote:
I couldn't think any recipe to reject a call if no callerID is
presented.
PrivacyManager and Zapateller are not an option, as the call will be
answered before I can drop it. I just want to silent drop the call:
no
callerID, no answer.
Any ideas?
I would imagine a simple gotoif followed by hangup would suffice
in psuedo code:
if (${CALLERID} == )
then
hangup
else
goto(incoming,s,1)
fi
I am not familiar with gotoif, but show application gotoif should
help.
Regards,
Adam
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RE: [Asterisk-Users] Wonderful Success with PAP2-NA
WHERE DID YOU GET THE PAP2-NA?!??!!? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, October 19, 2004 4:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Wonderful Success with PAP2-NA Finally got authorized to purchase some PAP2-NA's from Linksys's. Works like a charm with Asterisk. Web configuration has TONS of options and looks nice. Able to put line1 and line2 on seperate asterisk servers. Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4 line ATA for $100. -Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI for X-Ten Pro?
There is a setting in your sip.conf called mailbox. If you add this setting to your config it will send a message waiting signal to your soft phone. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Monday, October 18, 2004 11:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI for X-Ten Pro? Hi Folks, I shelled out for some licences for the X-Ten Pro phone so that we could use it whilst away from the office. Only problem seems to be that I can;t seem to work out how to make it tell me if I have VM without dialing the VM system. Any ideas? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicepulse down for anyone else?
They were for me.. But back up now.. brian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 18, 2004 1:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse down for anyone else? Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Project - IP Phone Sources
You have more options than you know. You could go with a channel bank if you want to keep support for the analog phones in the classrooms now(my school had them) or you could goto the next step with the sip phones. I have looked around and found a couple vendors to be fairly inexpensive. Check this link out: http://www.voip-info.org/wiki-VOIP+Phones Check under hardphones. It's a very good resource for the information your looking for. As far as the dialplan. It would take no time to build what your looking for and get everything setup. Got any questions feel free to drop me a email .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stewart M. Ives Sent: Friday, October 15, 2004 12:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Project - IP Phone Sources Hello, Background: Old to UNIX Linus, New to list. A techie Dad that supports local k-8 school that my kids go to. More background: Recently the school wanted to put phones in all the classrooms for teacher communications to/from the office. Another Dad in the telecom business spec'ed out a standard PBX with wiring, etc. Needless to say it was Expensive with a Captitol E. Anyway I started looking around at open source and found Asterisk. We currently have a complete switched network within the school (jsut replaced all hubs with switches) and have multiple PC's in each classroom as well as the front office. We also run RH Linux for our webserver, email server, file server, Websense server, and library software server. Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. Many thanks in advance. BTW, the school is: www.sainttheresaschool.org stew Stewart M. Ives SofTEC USA WebSite: www.softecusa.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FireFly SIP Registration Interval
SJPhone .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deon Rodden Sent: Thursday, October 14, 2004 3:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FireFly SIP Registration Interval I put FireFly on my moms computer, but ran into a problem. She went home and was able to place calls from it (using her headset and such). But, she could not receive calls. I figured out the problem was with the registration, firefly doesnt re-register often enough, so the connection gets stale and the NAT Device forgets about the connection, so no new incoming calls can be made. I put X-Lite on her computer and changed the re-registration interval from the default of 3600 to 60 seconds. Now I can call her anytime. But, theres choppiness on the line. Her ability to transmit/upload/send voice to me is bad, I hear choppiness and such. FireFly worked fine, no choppiness, same router, same connection. I tried X-Lite and FireFly on my laptop but both perform equally. I like the simplicity and interface of firefly, its nicer, anybody know of a way to change the sip registration interval? Anybody know of another program other than x-lite or firefly? One that doesnt have problems sending audio and one that allows you to change the sip registration interval? Thanks, Deon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Running Asterisk on Linksys Router
I run asterisk at my house on a linksys router. I have it sitting in the DMZ of the router so it acts like its outside. Works perfectly fine. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sent: Thursday, October 14, 2004 3:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Running Asterisk on Linksys Router At Astricon Mark mentioned that somone had Asterisk running on a Linksys Router. Anyone have more information on this? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
