Re: [asterisk-users] Had it with Dell Garbage - HP Question
You don't have to build Supermicro stuff yourself if you don't want to. Most Supermicro dealers do it for you if you buy all the parts from them. It's true that what your doing with Dell/HP is paying for emotional support. When it comes to PBX's you not getting any value paying for Dell/HP support. If your getting good stuff you should never need their hardware replacement warranty either. -Original Message- From: Jesse Molina [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 25, 2008 9:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question If you can't troubleshoot a hardware problem, then you should definitely not be thinking about this. Going with a support-yourself plan is not for everyone, especially if you don't have good hands-on hardware ability local to where the systems are. The cost savings can be significant enough that you don't' need to worry about third party support. Just buy some cold-spare systems. Out of eight servers, you can buy three spares and still have money left over -- assuming a three year life span for the systems and related support costs. This is true on 1U and 2U systems -- I'm not sure about larger stuff. After all, what are Dell/IBM/HP-Compaq going to do for you, other than replace the hardware? Nothing. They don't support your custom software and configurations, just hardware. Yank the hard drives, RAID controller, install in a spare system, and you're up and running again. Figure out what went wrong on the old system later. If it's under warranty, get it RMAed at your leisure. If it's not, you've got another two spare systems on the shelf, waiting there 24x7 just for you. Once again, it's not for everyone. If you don't feel comfortable with it, don't do it! It works for some businesses, not for others. It depends on who is supporting your servers. If IBM supports your servers, get IBM support. If you support your servers... then why are you paying them to do nothing??? Don't pay for emotional support. On Tue, Mar 25, 2008 at 06:55:10PM -0400, Al Baker wrote: ok - but, who do you call for HW problem ? HP has all levels of warranties all depending on how much $$ you want to spend. What do you do if you buy and install Supermicro ? HP also has 24x7 support center , again not for free, what do you do with Supermicro ??? I am really interested because I hear a lot of folks putting * on them but I never have worked on them while I have put in bunch of HPs really big boxes.. Thx for sharing your experience Matthew Gibson wrote: I've had good luck with these guys: http://rackmountsetc.com/ supermicro have never failed me yet. On Tue, Mar 25, 2008 at 6:03 PM, Jesse Molina [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you want barebones where you add your own processor, RAM, hard drives, and options, try SuperMicro brand servers. They are thousands of dollars less than the big (fat) names like IBM and HP/Compaq, but very good quality. I've built several clusters of computers with SuperMicro systems. They are great if you want to do barebones, clusters, or other special projects. It just takes a little more time to do the assembly work. Try newegg.com http://newegg.com for some sample pricing. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
Supermicro with hotswap bays and KVM card does the same thing. From: Darren Wright [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 26, 2008 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non-techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
That probably includes 5 years of support but still expensive. John Faubion wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? I actually work as a software engineer for a big telecom manufacturer to remain unnamed. I use Asterisk at home and I built a system for my aunt's real estate office mainly because she was quoted $83K+ over 5 years for a 12 station Toshiba key system. What on earth does that system do? open the office, make coffee, sweep the floors and ??? For such a small system there is no earthly reason for it to be 10 percent of that, even on a 5 year lease. Unless there are contract reasons she shouldn't even consider a lease either. I know that EVERYTHING is big in Texas, but that is nothing more than highway robbery. An NEC DSX with CF voicemail and e-mail integration wholesales for well under 3K, double that and add cabling . . . Well, you get the idea. John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
All arguments aside, I'll guarantee you MS OCS is much less stable than Asterisk/Linux, much more buggy, and will be for the forseeable future. It took M$ 5-10 years to get Exchange right. So in a few more years I'll have another look at it but I won't be a guinea pig for now. Sounds like things will be heading towards SIP/TCP in the future but if M$ made their implementation 'standard' it would be a precedent for them. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 11, 2008 6:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Microsoft Office Communications Server Right. Asterisk never crashes. Asterisk is completely solid. At the end, if you do not answer a call some else will!!! Three are not convincing enough. I think the following is more convincing: SIP/TCP will eat your babies! -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
I would rather stick needles in my eyes but that's just me. -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Sunday, March 09, 2008 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Microsoft Office Communications Server -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH1KVoDQNt8rg0Kp4RAgSfAJ0aXGIkKi6kGAjZK8TtSV2mMj79qQCdHhAS 1jZ9sjtsTJ3O1R9J3giztw8= =Mlnt -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra Park Softkey
This is what I have working topsoftkey5 type: speeddial topsoftkey5 label: Park topsoftkey5 value: ##70 topsoftkey5 states: connected -Original Message- From: Russell Brown [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 04, 2008 1:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Aastra Park Softkey Quoth: OCG Technical Support [EMAIL PROTECTED] Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's setup like so: softkey4 type: speeddial softkey4 label: *Park softkey4 value: #,700 softkey4 line: 1 softkey4 states: connected So the bottom left soft key does and says Park when connected. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which phones to use ??
U..not really. Aastra is right up there. IMHO they are better for several reasons. From: Michael Graves [mailto:[EMAIL PROTECTED] Sent: Saturday, March 01, 2008 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] which phones to use ?? When in doubt there is only one sure answerPolycom. Without a doubt the best functionality, performance and reliabilityeven in the lower cost models. Although the lesser models are still over $100. Michael --Original Message Text--- From: Rob Hillis Date: Sat, 01 Mar 2008 11:07:58 +1100 For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another gets worse and worse. randulo wrote: On Fri, Feb 29, 2008 at 1:12 PM, Agnello George [EMAIL PROTECTED] wrote: but it does not mention the phones that i need to use could i use any USB phone !!! ??? I would recommend you start by using free softphones like X-Lite, Gizmo project, Zoiper. Then, when you're ready, choose a hardphone by price and quality needed between Grandstream, Sipura, Polycom and Cisco not to mention Snom or Aastra that have a lot of models as well. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds.
Ok so I'm not going crazy then. I filed a bug report. http://bugs.digium.com/view.php?id=12093 -Original Message- From: Trevor Peirce [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 27, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds. shadowym wrote: A bit hard to describe. Using a SIP hardphone I log into my voicemail at which point Allison says you have x messages.. There are various other prompts that exhibit the same problem but that is one easy to explain and reproduce one. The problem is there is a slight 'pop' sound usually during the first syllable of each word. So the prompt sounds like y[pop]ou h[pop]ave t[pop]wo me[pop]ssages. If I dial *43 to do an echo test and Allison says you are about to enter an echo test. it's not there so it's only in certain modes this happens. Yes, this is something that has bothered me since I first started working with asterisk 1.2 way back when. It sounds to me like it's an artifact of appending multiple sound files together as it occurs at the beginning of each prompt that is played, or each digit when reading back caller id. I too see this with gsm, ulaw, and the new slin files. I know it happens with 1.2 and 1.4 on Sipura/Linksys ATAs. I just listened to the prompts on my Aastra 9112i and the pop is there too but not nearly as apparent as on the ATAs. I've got no idea where to even start trying to solve something like this, but I just wanted to respond that you're not the only one being bothered by it. Best regards, Trevor Peirce -- Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds.
This is something that has been bugging me for awhile. I have noticed it on multiple systems with different hardware, with and without zaptel cards, and various versions of Asterisk in the 1.2 and 1.4 branches. A bit hard to describe. Using a SIP hardphone I log into my voicemail at which point Allison says you have x messages.. There are various other prompts that exhibit the same problem but that is one easy to explain and reproduce one. The problem is there is a slight 'pop' sound usually during the first syllable of each word. So the prompt sounds like y[pop]ou h[pop]ave t[pop]wo me[pop]ssages. If I dial *43 to do an echo test and Allison says you are about to enter an echo test. it's not there so it's only in certain modes this happens. It's relatively minor and not something people generally complain about but it's definitely there and (at least to me) kinda makes the system seem to be a bit less polished sounding. I thought it had something to do with the prompts I am using but when I play back the actual files in a sound player I don't hear those pops. I tried various formats including gsm, ulaw etc. but it doesn't change anything. Anyone know if this is a documented issue and if so are there any potential solutions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High CPU load after upgrading to 1.4
Did you file a bug report? http://bugs.digium.com -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Thursday, February 21, 2008 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High CPU load after upgrading to 1.4 On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote: I currently have 1558 sip peers loaded in Asterisk and the current CPU load is 10% when no calls are being processed and no sip registrations. At first glance, I would think that maybe you have qualify=yes in each of your SIP peers, which is keeping Asterisk busy checking to see if the peers are responding or not. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voted most stable and easy to use phone?
I guess someone has to say it. Have you considered Aastra? You can argue about quality/features/functionality but I have set up both and the Aastra are definitely easier to configure and they reboot quicker. Nobody ever complains about the quality of sound or speakerphone on them either. From: John Faubion [mailto:[EMAIL PROTECTED] Sent: Thursday, February 21, 2008 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voted most stable and easy to use phone? A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. We just installed a dozen of the Polycom IP-330 phones. Initially out of the box I wasn't real sure about the decision to use them. The phones are very small and don't seem to have very many features. However in use they have been great. They don't waste a lot of desk space, they don't overwhelm the users and they seem to provide adequate information. They're easy to use and Polycom reliable. The speaker phone is still really good though I'm not sure it is as good as the 501/601 phones. I haven't really done a side by side comparison of that but I think the 501/601 has a better speaker phone. I can't see buying another GXP after using these. The difference in price just isn't worth the aggravation. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
How about a technical comparision. What makes the Rhino better than the Sangoma? On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal experience so I strongly disagree with that part of your argument. -Original Message- From: James Finstrom [mailto:[EMAIL PROTECTED] Sent: Friday, February 15, 2008 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b59f18311805637012918! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
How about a technical comparision. What makes the Rhino better than the Sangoma? On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal experience so I strongly disagree with that part of your argument. -Original Message- From: James Finstrom [mailto:[EMAIL PROTECTED] Sent: Friday, February 15, 2008 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b59f18311805637012918! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
Do you have a range of registration ports configured and forwarded through the firewall on the server end? Ie. 5060-5065 for example. On the Phone side you should forward 5060 to phone1 and 5061 to phone 2 etc. and configure the phones to use that port for registration. You may need to forward ports for the actual voice as well. 2 ports per phone so 1-10001 for phone1 and 10002-10003 for phone2. It's either that or mess around with STUN or Proxy servers or whatever. SIP+NAT=headache -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, February 02, 2008 8:23 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall The server is at a remote datacenter - no nat, no firewall, pure public IP. The phones are at home offices (i.e. DSL or Cable with Linksys-type firewall/routers). My initial testing was with a single SIP phone at the home office - and everything worked fine. But when I have two SIP phones at the home office, things start behaving badly. I understand the issue of phone-to-phone, where both phones are behind a nat at the home office - but that is not the issue I am having. My main problem is when I have two phones at the home office, the second phone cant register, and/or, you cant here the voicemail greeting when you try to check messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
Actively maintained or actively being broken and fixed with constant updates? Not something suitable for Production IMHO. Makes more sense for development and experimentation IMHO. -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Enterprise or Fedora? shadowym [EMAIL PROTECTED] writes: I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is only a yum install asterisk away! /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
I think CentOS 5 is better with Dual/Quad core than CentOS 4. I have no direct technical evidence of this. Just empirical from the Google oracle. From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Saturday, February 02, 2008 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel timer on Intel Dual Core servers Likewise here. The company I work for sells duo core boxes (though mostly with E1 cards) and we have no issue with timing. Chris Bagnall wrote: My question is if anyone else have seen this and if anyone has a possible solution? Nearly all of the boxes we've built over the last few months have been either Core2 Duo or Core2 Quad based and we haven't seen any timer issues. In most cases these are purely IP boxes (no PRI or TDM cards) with ztdummy. We've had plenty of meetme conferences over that time and I've not noticed any problems. I'll run zttest on a couple of them over the weekend and see what results we get, but certainly, there haven't been any complaints about quality during conferences. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. -Original Message- From: Matthew J. Roth [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Enterprise or Fedora? love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year old. It is also considered a bleeding-edge distribution and is a testing ground for new applications and features. As such, it has frequent updates (including the kernel) that aren't guaranteed not to break your system. If stability and long-term support are your goals, I recommend taking a look at CentOS http://www.centos.org/. It is a binary-compatible clone of Red Hat Enterprise Linux that's free and has a very long support period. It's basically RHEL without the paid support contract. My migration to CentOS was painless, because the file system and configuration are practically identical to those of Fedora. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large issue - having trouble diagnosing.
H,
Using some brand new obscure Asterisk distribution, TE110P card, Grandstream
phones.
Can't imagine why you would be having problems
{/sarcasm off}
From: Cameron Hissey [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 20, 2008 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Large issue - having trouble diagnosing.
Hello,
I am having a lot of trouble with my deployment of Asterisk. I am running
the PBX-In-a-flash turnkey of Asterisk and ever since deployment I have had
many different problems. I have managed to get all issues sorted out as I go
along, until this one that randomly began last week.
We are using Grandstream GXP 2000 Handsets in the office, and at TE110P card
to interface to our ISDN OnRamp10 connection (10 Channels of PRI).
The problem arising seems to happen roughly 4minutes into a call. Basically
all of a sudden the caller just starts to no longer be understood (sounds
like morse code, only milliseconds of voice packets getting through in
either direction). naturally this could be a number of non-asterisk related
things such as a carrier fault, bad network wiring (even more possible as we
are using PoE), even badly configured QoS. However things being as they are
my boss has taken it upon himself to absolve himself of any possible blame
for any system that he manages (everything but the asterisk box) and lumped
it all on me in such a way that its basically my job if i cannot get this
working. With all of this, i need to do everything i can to rule out the
Asterisk box, so i can go back to him with confidence and clear asterisk of
any wrongdoing.
Has anyone here ever heard of this sort of problem, and if so did you find a
solution? If not, what steps would you recommend i take to diagnose the
issue and rectify it as quickly as possible?
Thankyou very much,
Cameron Hissey
___
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[asterisk-users] Asterisk on ClarkConnect
Has anyone tried installing Asterisk on ClarkConnect? It looks like ClarkConnect runs on RHEL so it should work if they haven't modified it too much. It appears that ClarkConnect is working on adding Asterisk and integrating it into their GUI but until then I'd also be interested in trying to use FreePBX. Anyone? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Delay in Audio Over Analog
What are you using for a PSTN gateway? From: Brian Alexander [mailto:[EMAIL PROTECTED] Sent: Monday, December 31, 2007 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] One Way Delay in Audio Over Analog I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
Yes did all that. I've configured sendmail before so know the basics. I tried modifying sendmail.mc and creating the sendmail.cf file and also tried modifying sendmail.cf directly. I always restart sendmail after changes. Would I need to create a noreply mailbox in sendmail perhaps? What creates the asterisk mailbox? Does that happen when I make asterisk? Maybe there are some clues in that script somewhere. -Original Message- From: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED] Sent: Thursday, December 20, 2007 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path Tilghman Lesher wrote: On Wednesday 19 December 2007 17:44:15 shadowym wrote: I had high hopes for this solution for unfortunately it's not working. Did exactly as you specified but return path is still [EMAIL PROTECTED] even though [EMAIL PROTECTED] in voicemail.conf :( Did you restart Sendmail? It doesn't pick up changes to its config file otherwise. And if you modified sendmail.mc instead of sendmail.cf, don't forget to regenerate sendmail.cf -- something like the following: cd /etc/mail; cp sendmail.cf sendmail.cf.todaysdate; m4 sendmail.mc sendmail.cf should work, followed by /etc/init.d/sendmail restart Mojo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
Thanks for the info Tilghman, I had high hopes for this solution for unfortunately it's not working. Did exactly as you specified but return path is still [EMAIL PROTECTED] even though [EMAIL PROTECTED] in voicemail.conf :( -Original Message- From: Tilghman Lesher [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 19, 2007 6:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path On Wednesday 19 December 2007 01:11:25 Johansson Olle E wrote: 19 dec 2007 kl. 01.07 skrev shadowym: Unfortunately that only changes the from field. So if you were to reply to the email that is the one Outlook would use. The receiving mail system looks at the return path in the header of the email to determine if it is valid. serveremail and fromstring do not change that. Again, the return path in the email is set to [EMAIL PROTECTED]. I can easily change mydomain.com in sendmail but cannot figure out how to change asterisk. Sendmail has a notion of trusted users that are allowed to change the envelope sender's address. Your Asterisk process userid propably does not belong to that group. Add it to the group in the wonderfully elegant and simple sendmail configuration and change the mailcommand in voicemail.conf so that you specify another sender. The line to do this is Tasterisk in sendmail.cf or in sendmail.mc: define(`confTRUSTED_USERS', `asterisk')dnl -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to change sendmail return path
Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
servermail= changes what shows up in the from section of the email. It doesn't change what shows up in the email header which is what the mail system looks at as the REAL return path. -Original Message- From: Mark Michelson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 18, 2007 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 The option you are looking for is called serveremail. By default, if this is not set, it will be set to asterisk. Set this in the [general] section of voicemail.conf. Mark Michelson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
Unfortunately that only changes the from field. So if you were to reply to the email that is the one Outlook would use. The receiving mail system looks at the return path in the header of the email to determine if it is valid. serveremail and fromstring do not change that. Again, the return path in the email is set to [EMAIL PROTECTED]. I can easily change mydomain.com in sendmail but cannot figure out how to change asterisk. -Original Message- From: Forrest Beck [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 18, 2007 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path Have a look at serveremail = [EMAIL PROTECTED] and fromstring = The Asterisk PBX in voicemail.conf. On Dec 18, 2007, at 2:28 PM, shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
I would rather the Developers spend their precious time improving the stablilty and reliability than creating a smooth upgrade process. Not that I don't think it is at least as reliable and stable as 1.2 right now. It seems to be for me in a low call volume environment. A PBX should be looked at as more of an appliance than an application server IMHO. You shouldn't have to upgrade it unless it was inadequate to begin with. If that is the case you should be doing an install of 1.4 from scratch anyways. Just my opinion. -Original Message- From: Phil Knighton [mailto:[EMAIL PROTECTED] Sent: Monday, December 17, 2007 4:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Hello As a person who is somewhat a newbie to Asterisk, I have been given the task of preparing our 1.2 installation for upgrade. The thing that has slowed me down is some of the gaps in information on the upgrade process. What's on the Wiki might make complete sense to both experienced Linux users, and Asterisk users but as someone who is feeling there way through - it's a bit daunting! Considering how important a phone system is to a business, I'm loathed to rush the upgrade through and have instead opted to install 1.4 on a different box, and port our existing setup over to it. This is a time consuming process and has taken quite a low priority. As Olle says - 1.2 works just fine. Personally speaking, the upgrade process has to be even easier if people are going to jump for it. Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: 15 December 2007 10:57 To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
I do wish Digium or whoever tests this stuff had a more reliable way of testing software releases rather than relying on feedback from the community. Fonality, for example use what they call a hammer which sounds to me like a bunch of servers running various stress tests on the software to try break it. -Original Message- From: Atis Lezdins [mailto:[EMAIL PROTECTED] Sent: Monday, December 17, 2007 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! On 12/15/07, Johansson Olle E [EMAIL PROTECTED] wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! We have switched to 1.4 some half year ago, and main motivation was some stability issues with 1.2 (and few new features), so having 1.4 for us means - we're actually having support - we can post bugs to Mantis, and got them solved. Our migration is not yet completely over, last step is getting rid of AgentCallbackLogin, that we plan to do in beginning of next year. However 1.4 since release have had some serious changes that blocked our planned upgrades - for example some memory corruption that raised between 1.4.10 and 1.4.12 that was very hard to track down. This shows that having 1.4 in bugfix-only state is not actually working that good - we have to test each new release very carefully. In total 1.4 have helped us to get rid of twice-per-week crashes we experienced on 1.2, so i would call it more stable than 1.2. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Most Stable version of Asterisk
1.4.15 on CentOS 5.1 is running smooth as silk for me. From: Jai Rangi [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 11, 2007 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Most Stable version of Asterisk Hello, I tried to install the asterisk 1.4.15 and I am not able to start it. I get the segmentation fault error. What might be wrong, where I can look for a clue. Also could some one PLEASE suggest the most stable version of asterisk. -Jai ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
There are probably a half dozen or more software apps that can do this. Most are free last time I checked. Google is your friend. From: Michael Melia Jr. [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 05, 2007 8:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk SIP Microsoft Outlook Integration Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Thanks, Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Shared line appearance phones?
That would be VERY much appreciated Russell, There seems to be a lack of info and the accompanying confusion/misinformation about this. -Original Message- From: Russell Bryant [mailto:[EMAIL PROTECTED] Sent: Friday, November 30, 2007 4:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Shared line appearance phones? Mark Wiater wrote: I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several Polycom IP430's. Might you be willing to share some specific configurations for such a situation? There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I have on my to-do list to spend a week with an SLA test environment and coming up with an extensive set of examples of the different ways it can be used. I will post something to this list when that is available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium and Asterisk
How is buying Digium the right thing to do? It is not like they are a true open source company. More like proprietary that likes to use the open source community to test. If you buy with your heart and not your head I pity your customers! From: Steven [mailto:[EMAIL PROTECTED] Sent: Monday, November 26, 2007 4:34 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium and Asterisk I only had 2 minor issues with Digium cards, and both were replaced with advanced shipping. I am very satisfied with all of my Digium cards. I also believe that supporting asterisk via a Digium purchase is the more right thing to do. Note: I have only used these in Dells with no motherboard issues. 1850, 1950, 2850 and 2950. -- -- Steven http://www.glimasoutheast.org Marco Mouta [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I got one of this boards and I got it successfully replaced by Avanzada7 (Digium official reseller) immediately. On Nov 24, 2007 6:46 AM, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Actually if you rule out all the clone tormenta cards (nothing wrong.. but very dated design... I wouldnt buy one today) the Digium cards aren't too expensive. Those tormenta cards are the ones you see for $300-400 typically. Some people like Digium others Sangoma. Personally I'm a Sangoma man. Some people report certain main boards and Dell servers aren't compatible with some digium cards. According to a post here on the mailing list someone from Digium implied that they will replace cards with these conflicts with newer model card that does not have these conflicts... your millage may vary I don't believe that forum posting was made in any official capacity but I also doubt that Digium would not do something to correct an issue for an item under warranty. On Nov 22, 2007 8:03 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. _ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchvox Space Requirements
I would get at least 4Gig. It's very inexpensive now. With wear management, the more space you have left over the longer the Flash will last. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Friday, November 09, 2007 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Switchvox Space Requirements Andres wrote: Can anybody give me a rough idea how much disk space is requiered for a typical install? I want to install it in a system with solid state storage and I don't want to buy more than I need. Would 1GB be enough? I don't believe so, I would expect that 2GB would be plenty though. Also keep in mind that there is no 'typical' install, because you don't get to make any choices during the installation... there is only one kind of install :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC and zaptel-1.4.5.1
There is a bug in Zaptel 1.4.5.1 that prevents other Echo Can's from being selected. http://bugs.digium.com/view.php?id=10555 Use 1.4.6 or 1.4.4 or edit the source yourself. -Original Message- From: marcotasto [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 24, 2007 9:08 AM To: asterisk-users Subject: Re: [asterisk-users] OSLEC and zaptel-1.4.5.1 Hi Alan. I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the 1.4 version and I have had the same problem. Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV is defined by default and this prevents the echo selection from zconfig.h. I've solved changing the first part of Makefile.kernel26 (in the zaptel directory) this way: ifndef ECHO_CAN_NAME ECHO_CAN_NAME := OSLEC endif This forces the compiler to include OSLEC as echo cancellation engine (probably there is a better way but I don't know it). I've then rebuilt zaptel and installed through normal make procedures. To be able to modprobe it I've then copied the oslec.ko file build by the OSLEC distribution in the kernel driver directory (my own is /lib/modules/2.6.18.8-0.5-default/misc and it's where zaptel drivers are installed). I've then run the depmod command to regenerate the modules dependencies. I'm now able to modprobe zaptel and to have oslec automatically installed as you can see below: lsmod | grep zaptel zaptel12 6 zttranscode,wctdm oslec 23332 1 zaptel crc_ccitt 6272 1 zaptel I hope this could help you. Best regards, Marco Signorini. Hi all, After reading great things about the OSLEC Echo Canceller (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of people who have tried it on a recent Trixbox thread (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-ec ho-problems), it sounds like it is the bees knees for sorting out echo problems with cards like the x100p. Has anyone managed to get oslec to work with recent zaptel and kernel (I'm running 2.6.23)? Lots of information below. Comments/suggestions welcome. Having followed the instructions on the oslec site, and ensuring the patch for zaptel takes O.K (I manually installed the patch into the zaptel source tree just to make sure). I can build the oslec module, and build a patched zaptel-1.4.5.1-oslec without any compilation issues. However when I reload the system during boot-up dmesg tells me: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.5.1 Zaptel Echo Canceller: MG2 Zaptap registered 'sample' char driver on major 33 (This means the patch went in O.K.) ACPI: PCI Interrupt :00:08.0[A] - GSI 17 (level, low) - IRQ 22 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Wildcard X100P Notice the choice of echo canceller If I look at what modules are installed: # lsmod Module Size Used by zttranscode 6280 0 ztdummy 3432 0 wcfxo 9760 0 zaptel200120 7 zttranscode,ztdummy,wcfxo crc_ccitt 1792 1 zaptel No oslec :-( In my kernel modules/misc directory I have: -rw-r--r-- 1 root root 10727 2007-10-24 14:44 oslec.ko -rw-r--r-- 1 root root 65372 2007-10-24 14:41 pciradio.ko -rw-r--r-- 1 root root 91321 2007-10-24 14:41 tor2.ko -rw-r--r-- 1 root root 18901 2007-10-24 14:41 torisa.ko -rw-r--r-- 1 root root 12605 2007-10-24 14:41 wcfxo.ko -rw-r--r-- 1 root root 15989 2007-10-24 14:41 wct1xxp.ko drwxr-xr-x 2 root root 4096 2007-10-24 14:41 wct4xxp drwxr-xr-x 2 root root 4096 2007-10-24 14:41 wctc4xxp drwxr-xr-x 2 root root 4096 2007-10-24 14:41 wctdm24xxp -rw-r--r-- 1 root root 41046 2007-10-24 14:41 wctdm.ko -rw-r--r-- 1 root root 32882 2007-10-24 14:41 wcte11xp.ko -rw-r--r-- 1 root root 45804 2007-10-24 14:41 wcte12xp.ko -rw-r--r-- 1 root root 16527 2007-10-24 14:41 wcusb.ko drwxr-xr-x 2 root root 4096 2007-10-24 14:41 xpp -rw-r--r-- 1 root root 81616 2007-10-24 14:41 zaptel.ko -rw-r--r-- 1 root root 8270 2007-10-24 14:41 ztd-eth.ko -rw-r--r-- 1 root root 5530 2007-10-24 14:41 ztd-loc.ko -rw-r--r-- 1 root root 5297 2007-10-24 14:41 ztdummy.ko -rw-r--r-- 1 root root 11687 2007-10-24 14:41 ztdynamic.ko -rw-r--r-- 1 root root 8639 2007-10-24 14:41 zttranscode.ko My /etc/zaptel.conf is: loadzone=uk defaultzone=uk fxsks=1 My /etc/asterisk/zapata.conf is ; Zapata telephony interface ; ; Configuration file [channels] ;Hardware defaults for the x100p card ;usecallerid=yes ;hidecallerid=no ;callwaiting=no ;threewaycalling=yes ;usedistinctiveringdetection=yes ;transfer=yes ;usecallingpres=yes ;callwaitingcallerid=yes ;cancallforward=yes ;callreturn=yes echocancel=yes echotrainingwhenbridged=no ;echotraining=400 rxwink=300 ; Atlas seems to use long (250ms) winks ;cidsignalling=v23 ; Added for UK CLI detection ;cidstart=usehist ; After patching the driver from here : ;
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
Or your could use a touch screen with Flash Operator Panel. Just a suggestion out of left field. -Original Message- From: Russell Brown [mailto:[EMAIL PROTECTED] Sent: Friday, October 19, 2007 1:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Receptionists Phone suggestions? (Not Snom370) Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details below if you/re interested). I've verified this problem with Snom who's response is that the receptionist should answer all of the incoming calls before trying to do a transfer - That's just Bonkers! So... any suggestions? Details of Snom 370 problem for the record: Snom370 gets a Call (Call A). Snom370 answers Call A. Call A wants to be transferred to Phone C. Snom370 has another call ringing (Call B). Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B still ringing. Snom370 Dials Phone C (Call C). Snom370 talks to Call C. Snom370 presses TRANSFER. The display shows: CallA CallB The soft keys now show and . Pressing them does nothing. When the TRANSFER button is pressed again, CallA is connected to CallB (the original caller is now talking to the previously unanswered party) not what one wanted to happen! It's not difficult to see why my client is throwing their toys out of the pram and I'm going to have to replace the Snoms at my expense :-( -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
It's just FOP which works well. Dependent on the quality of touch screen obviously. I haven't spend any time with FOP using Touch screens myself but I'm sure others here have. There was a thread a few days ago that got into it a bit. -Original Message- From: Mike Clark [mailto:[EMAIL PROTECTED] Sent: Friday, October 19, 2007 8:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370) shadowym wrote: Or your could use a touch screen with Flash Operator Panel. Just a suggestion out of left field. snipped a bunch shadowym: Do you have a specific setup w/touchscreen that you have deployed and that works well? Thanks, Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
Ok so you use templates. I understand that. The problem is some people on here seem to be claiming they type it all in from scratch in like 3 minutes. -Original Message- From: Lenz [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 17, 2007 12:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What web GUI are people happy with? Well, most of the configuration (but the dialplan) can be kicked up pretty fast by preparing a block (eg for a SIP extension) and then pasting it over with minor modifications. I usually keep the original demo config files around, that already include most options, and use them as a starting point. About the dialplan, I have a few templates around and usually use them as a starting point for more complex things. Easy things come usually easily and usually work the first time I try them. Some of them I posted to astrecipes.net or voip-info as well so I have them handy all of the time. OK, I'm on this list since 2004 so maybe I'm a bit biased, but it really works fine for me. l. On Wed, 17 Oct 2007 05:23:14 +0200, shadowym [EMAIL PROTECTED] wrote: That seems quite fast actually. I still have a hard time believing it is that easy. I look as relatively straight forward configurations that do call queues, voicemail to email, followme, ivr and various other things that most companies want and your looking at hundreds of lines a lot of which are not redundant with just extensions etc. changed. I just don't see how anyone could set that all up in 30 minutes. Unless of course your cut and pasting templates. Something nobody seems to claim they use. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
Ok Thanks, I guess I'll have to give it a shot. I just assumed it would be more work than 30minutes (after the initial learning curve) for a moderately complex dialplan.. -Original Message- From: Erik Anderson [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 17, 2007 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What web GUI are people happy with? On 10/17/07, shadowym [EMAIL PROTECTED] wrote: Ok so you use templates. I understand that. The problem is some people on here seem to be claiming they type it all in from scratch in like 3 minutes. Just call me out if you feel the need to. Please don't try and hide behind the some people on here type of comments. Call me out directly if you feel the need. I can take it :-) So...I don't feel the need to prove myself to you. I have a fairly good grasp on the conf file syntax, and with a well-thought out and well documented goal, it's not unreasonable for me to say that I can type out a config from scratch in 30 minutes. After working in vim for as long as I have, you learn to use the many shortcuts that it provides for text manipulation, copy buffers, moving blocks of code around, etc. I also use a syntax highlighting rule file for asterisk configs, so any typos I make are immediately evident. It's really remarkable how this discussion has turned into a pissing match. I could really care less if you have a hard time believing my statements. I'm not trying to push CLI on you or anyone. Yes - I recommend that people give it a try before going to a GUI, but I fully recognize that vanilla asterisk text configuration isn't for everyone. -Erik P.S. By the way - don't misquote me. I said nothing about laying down a config in 3 minutes. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone having any luck with Bluetooth?
I have read all the wiki's and blogs and how to links about Bluetooth but so far no luck. I can confirm that CentOS5 sees my Bluetooth adapter and my cell phone. No Joy on Asterisk 1.4. The information out there is kind of confusing as there is a lot of outdated info sometimes referring to software no longer actively developed. What I think I have managed to conclude is that there is a Bluetooth module for Asterisk 1.4 that supposedly works but it's not part of any released branches so I will have to use a development branch. That is ok but I still can't get it to work. I want to use my Cell phone as a secondary trunk for a business so I need it to work reasonably well. Does it (if and when I get it working)? Is there a good (up to date relevant) how to somewhere? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on USB Flash?
Size/Speed/write cycles have gone way up, price has gone way down. More common than CompactFlash and no need for an adapter. So is it feasible to run an Asterisk server on something like this? With a MTBF of 1million write cycles coupled with dynamic wear management on a 4Gig USB drive, lifetime is a non-issue. Just wondering how well it works, if it works. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
So nano just makes things too easy for you? -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Monday, October 15, 2007 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What web GUI are people happy with? I use vi. Not sure if it has a web interface yet. PaulH On Tue, 2007-10-16 at 00:51 +0200, Dovid B wrote: None. Asterisk vanilla is the best IMHO. - Original Message - From: Anciso, Roy To: asterisk-users@lists.digium.com Sent: Monday, October 15, 2007 7:28 PM Subject: [asterisk-users] What web GUI are people happy with? Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
I don't do text editing so please indulge me. Why would someone want to do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of fancy things like call
queues, parking, forwarding, followme, voicemail to email etc. etc. how
practical is it to type all this in by hand making sure to get ever single
space, ., ,, {}, [] etc. exactly right which NEVER happens. So then
you have to spend more time debugging the conf files.
Even with a bunch of pre-made templates it seems like an awful lot of
unnecessary heavy lifting when a GUI can make it so much easier and
efficient.
-Original Message-
From: Lenz [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 16, 2007 12:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
If what I see actually deployed is what people are happy with, I'd say the
lion's share surely goes to FreePBX. Another GUI that lots of people like
when they try it is Voiceroute Druid, but it's a commercial product. Of
course, no GUI gives you the amount of flexibility that a text editor buys
you, provided you know what you're doing. :)
l.
On Mon, 15 Oct 2007 19:28:48 +0200, Anciso, Roy [EMAIL PROTECTED] wrote:
Just wondering what web GUI people like for asterisk. I installed
asterisk from source and I was looking at possibly installing web GUI
for system management. So far freepbx.org looks promising anybody else
have any suggestions.
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-723-1690
[EMAIL PROTECTED]
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
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Re: [asterisk-users] What web GUI are people happy with?
So how long would it take you to vi a 20 extension office with custom
dialplan involving a medium level of complexity? Including time to debug
etc.
-Original Message-
From: Erik Anderson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 16, 2007 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
I don't do text editing so please indulge me. Why would someone want to
do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of fancy things like call
queues, parking, forwarding, followme, voicemail to email etc. etc. how
practical is it to type all this in by hand making sure to get ever single
space, ., ,, {}, [] etc. exactly right which NEVER happens. So
then
you have to spend more time debugging the conf files.
Even with a bunch of pre-made templates it seems like an awful lot of
unnecessary heavy lifting when a GUI can make it so much easier and
efficient.
This is *very* much a to each their own issue. You say that a web
GUI is more efficient - I say that vi is more effecient. You say that
using a text editor is more error-prone - I say that a web GUI is more
likely to mess things up in a difficult way to troubleshoot.
Use what works for you and don't worry about it.
:-)
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Re: [asterisk-users] What web GUI are people happy with?
I agree that a GUI can never be as flexible and efficient as raw script.
That is not my point. My point is that the more complex things get the more
an abstraction layer of some sort makes sense. It's the same reason people
use C code instead of doing everything in assembly language. Not as
efficient but much less work.
I fail to see how anyone can write all configuration scripts for even a
relatively simple dialplan for an Asterisk server in 3 minutes! You must be
talking about using templates yes?
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 16, 2007 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
I don't do text editing so please indulge me. Why would someone want to do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of fancy things like call
queues, parking, forwarding, followme, voicemail to email etc. etc. how
practical is it to type all this in by hand making sure to get ever single
space, ., ,, {}, [] etc. exactly right which NEVER happens. So then
you have to spend more time debugging the conf files.
Even with a bunch of pre-made templates it seems like an awful lot of
unnecessary heavy lifting when a GUI can make it so much easier and
efficient.
You're welcome to do it however you like. But please don't suggest that
using a GUI will make things more efficient. Someone with experience
scripting can easily write a system to generate a well-formed, valid .conf
file, with appropriate comments. I, for one, have done this.
The reason many seasoned Asterisk admins prefer using the .conf files
instead of using a GUI is that no GUI can possibly conceive of every way to
do something. So, at some point, if your PBX does anything interesting,
you're going to have to integrate your changes with what the GUI generated.
And not let the GUI stomp on the changes. But make sure everything will be
in contexts that can access what it should, and not access what it
shouldn't.
Now, as far as how practical it is to create the dialplan by hand, I can
tell you that it only takes about 2-3 minutes to full configure such a
simple PBX as you described. Most GUI systems take far longer than that to
install, much less configure. Also, I can more easily manage systems
remotely via SSH than through many of the GUIs out there.
So, as I said before, do whatever works best for you. But please don't
insinuate that editing configuration files cannot be a good idea.
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Re: [asterisk-users] What web GUI are people happy with?
That seems quite fast actually. I still have a hard time believing it is that easy. I look as relatively straight forward configurations that do call queues, voicemail to email, followme, ivr and various other things that most companies want and your looking at hundreds of lines a lot of which are not redundant with just extensions etc. changed. I just don't see how anyone could set that all up in 30 minutes. Unless of course your cut and pasting templates. Something nobody seems to claim they use. I don't see how using SSH console with something like PuTTY can influence the decision. I access all my GUI's through SSH port 22 using PuTTY tunnel which takes 30 seconds to configure for port 80 and 4445. -Original Message- From: Erik Anderson [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 16, 2007 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What web GUI are people happy with? On 10/16/07, shadowym [EMAIL PROTECTED] wrote: So how long would it take you to vi a 20 extension office with custom dialplan involving a medium level of complexity? Including time to debug etc. Well - there's a large amount of subjectivity in your question, but perhaps I'll answer with not long. I don't know - 20 sip extensions, maybe 5 minutes. Probably another 30 for the dialplan and debugging. My point still stands - use what you're comfortable with. I spend the vast amount of my day working through an SSH console into various linux servers, so it would only make sense that for me (and many other CLI geeks), it doesn't make sense to use a GUI. I actually get a little put out when I have to switch over to my browser or another GUI tool to get things done. So - the CLI is what works for me. I'm not going to push that on you or anyone as the definitive best management tool for asterisk. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
I hope I am not opening a can of worms here but IMHO there is ABSOLUTELY NO REASON TO USE SCSI anymore! For sure not for this application but most other things too. SATA is mature now, does command queuing, and works well on 2.6 kernels. Oh, there is the issue of cost as well. -Original Message- From: Matthew J. Roth [mailto:[EMAIL PROTECTED] Sent: Monday, October 15, 2007 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk? Raúl Gómez C. wrote: Thinking about my original post, I was reluctant of installing my PBX on a shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual Core CPUs @2GHz (4 totals cores) and 4GB RAM which serves as Domain Controller and File Server (Samba), central backup server (Bacula with a LTO2 external tape drive), it has dual NIC in a bonding alb mode and redundant PSU (each one connected to a different UPS). It has a PCI slots in which I can install my Sangoma Remora A400D card. Or its better to install the PBX on a dedicated system? Let me know your opinions! Raúl, The short answer is yes, it's better to install Asterisk on a dedicated system. The long answer is that you could probably get away with it, but if you have problems you'll be dealing with a lot of variables during troubleshooting. When that time comes, I wouldn't be surprised if the first piece of advice people give you is to offload the non-VoIP related tasks from the server. For 35 simultaneous calls, I'd recommend a dedicated server with a 3.0 GHz dual-core CPU, 2 GB of RAM, and fast SCSI disks. In my experience, the FSB can be just as important as processor speed so keep that in mind as you lay out your budget. You should be able to buy something from Dell with redundant power supplies (and other convenient features like a remote access controller) for an affordable price. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Whatever your many reasons, using that stuff for Asterisk is a waste of money but go crazy if you want! -Original Message- From: Shaw Terwilliger [mailto:[EMAIL PROTECTED] Sent: Monday, October 15, 2007 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk? shadowym wrote: I hope I am not opening a can of worms here but IMHO there is ABSOLUTELY NO REASON TO USE SCSI anymore! For sure not for this application but most other things too. SATA is mature now, does command queuing, and works well on 2.6 kernels. Oh, there is the issue of cost as well. This is just not true. If you want the best performing drives out there today, you'll be using SAS (Serial Attached SCSI) or Fibre Channel. There are still 3.5 LVD SCSI drives (the old parallel style) that beat the pants off the fastest competing SATA drives because they spin at 15K RPM and have longer MTBFs. Yes, these drives cost more than SATA, but there are many reasons to use them. -- Shaw Terwilliger [EMAIL PROTECTED] SourceGear LLC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
That's kinda high then. I wouldn't be happy about that either. You shouldn't be over 30% ever for anything real time. Instantaneous spikes can really start to make your life miserable at that point. -Original Message- From: Erik Anderson [mailto:[EMAIL PROTECTED] Sent: Friday, October 12, 2007 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk? On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I wouldn't be too happy about a system with a loadavg of 3. The system he mentioned had 8 cores, though. So a load average of 3 is less than 50% usage. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Embedded Distro
Astlinux does seem to be growing cob webs a bit. Askozia doesn't support Zaptel cards in the GUI and not sure if it is possible to configure them manually. There is no Voicemail storage mechanism yet. It's still very basic but a nice start. -Original Message- From: Michael Graves [mailto:[EMAIL PROTECTED] Sent: Sunday, October 07, 2007 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] New Embedded Distro Hi All, A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest build of Askozia PBX on a small system fors testing. The GUI looks nice. I' not certain if I want v1.4 in production as yet. If that proves to be the case then Askozia looks like a candidate to replace Astlinux, which is v1.2 and has essentially no GUI. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
I disagree with any argument for or against Digium in support of Asterisk as much as I do for or against Sangoma or Rhino or one of the Chinese knock offs in support of Asterisk. Digium uses the open source community to create better commercial software products and their licensing policies reflect that whereby they own the code anyone contributes so it's not like they don't benefit if you don't buy their hardware. I say buy the best product for the job whether that be Digium or Sangoma or Rhino or whatever. Nobody should feel obligated to buy one over the other for any reason whatsoever. Nobody is saying that the community should not be grateful for what Digium has and continues to do. Nobody should be saying that the community should be expressing that gratitude by buying Digium hardware either. Just as the community should not be asking Digium to remove their licensing restrictions to express their gratitude for the thousands of individuals not affiliated with Digium that test and document and file bug reports and submit patches for Asterisk every day. My 2bits. -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Friday, October 05, 2007 8:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too) Brian West wrote: Sangoma has contributed to Asterisk in the past and they still do. They also have contributed to Yate, FreeSWITCH and various other software that is capable of using their hardware. This argument of Digium vs Sangoma is very emotional for some. I see it as competition is good and drives innovation. Digium can't take every bit of credit for Asterisk, you have to remember the community has a large part in making Asterisk as popular as it is. I know their is hostility directed at anyone that uses non-Digium hardware by some folks and their shouldn't be. Its an open market and an open platform. Rhino makes hardware that plugs into zaptel but yet I don't see their drivers in the zaptel repo... I don't see many of the third party hardware drivers in the zaptel repo. Not to ignite any fires, but I don't think I've *ever* knowingly received a patch to libpri or chan_zap from them. And I've fixed a few protocol related bugs in libpri for people with Sangoma cards. It'd be nice if they at the very least supported the protocol stacks and zaptel channel driver they use to make money off their cards. Matthew Fredrickson /b On Oct 5, 2007, at 7:51 AM, Steve Murphy wrote: Oh, Julian, I'd imagine what I'm about to say will fuel some flames! Here's a fairly powerful argument for all you asterisk users, as to why you should purchase a Digium product vs. a Sangoma: Because Digium uses a chunk of the purchase money to support Asterisk. And Sangoma DOES NOT. Digium employs several developers specifically to maintain and improve Asterisk. Sangoma DOES NOT. While they may maintain and improve their own versions of the various drivers, THEY DO NOT SHARE THEIR SOFTWARE. Matt F. told me last week we haven't seen ANYTHING from them for a LONG TIME, with respect to the zaptel drivers. If they have been contributing patches, they are disguising their association with Sangoma. Don't get me wrong. I AM a Digium employee! A software Developer to be specific, an Asterisk developer to be precise. So, I AM highly biased towards Digium! Digium has a harder job than Sangoma with respect to Asterisk. While Digium takes a chunk of its revenue, and uses it to maintain and improve Asterisk (not just the drivers), Sangoma doesn't, and it gives them a competitive edge. So, for all you folks who have bought Digium, I personally thank you! You have helped Asterisk, and you have personally helped ME. If you have long-range business or interest in Asterisk, you are indirectly contributing to its growth and improvement when you buy Digium products services. murf -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Like fixing the poor design of the TDM400P and TE110 with the newer cards that advertise VoiceBus. For a company that supposedly embraces the open source philosophy I don't think Digium has been very forthcoming with what they are doing so they should not be surprised by any apparent lack of understanding from the community. It seems to me like it's been and continues to be a one way street. The dual licensing scheme is indicative of that as well. -Original Message- From: Russell Bryant [mailto:[EMAIL PROTECTED] Sent: Friday, October 05, 2007 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too) Brian West wrote: Sangoma has contributed to Asterisk in the past and they still do. Which contributions are you talking about, exactly? I know that they paid someone to write app_dictate a couple of years ago, but that is the only thing I can think of that has come through since I have been involved (for a little over 3 years now). In that same time frame, the number of bug fixes and new things coming from Digium is many, many hundreds, and likely in the _thousands_. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Because there is still old hardware in the pipeline at distributors and resellers. They may even still be manufacturing some of that old hardware so there is probably a lot of unrealized money involved. So in that sense I can't blame them for being a bit hush hush about it's short comings. Digium is advertising the new hardware improvements but at the same time they are careful not to talk about the shortcomings of older hardware. Just look for the discussions about VoiceBus on their site. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Friday, October 05, 2007 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too) Jared Smith wrote: On Fri, 2007-10-05 at 11:32 -0400, Steve Totaro wrote: I used to buy Digium products until they let me down with all kinds of quirky behavior with regards to echo, clicks, incompatible motherboards and IRQ issues. (Let me take off my Digium hat for a minute and speak as a community member, and not as a Digium employee. The opinions expressed below are my own, and not Digium's.) I'll be the first to admit that Digium has made some design mistakes on some of their cards in the past. And since I've been doing Asterisk consulting and also teaching Asterisk Bookcamp classes for the past two and a half years, I've seen Digium hardware installed on a wide variety of motherboards. I'm happy to report that I haven't had any motherboard or IRQ incompatibility with any of Digium's newer cards that use their VoiceBus technology. In some ways, I think Digium hasn't said enough about their efforts to fix the problems of the past and to make sure that their products are rock solid. So if anything, I've gotta stand up for Digium a little bit here and say Yes, they've had problems in the past. But you shouldn't discount the effort they're making to change that. That being said, I've also had the opportunity to spend a lot of time using hardware from most of the other board vendors. For the most part, they've worked OK for me and my clients, but they aren't perfect either. The biggest problems have been related to drivers (either drivers that are buggy, or won't work with the latest version of Zaptel or the kernel). Obviously in a perfect world the third-party drivers would be integrated right into Zaptel, and the zaptel drivers would all be pushed upstream into the Linux kernel. Until that happens, we're going to continue to have these sorts of problems to a certain extent. (I also have a philosophical problem with certain of the smaller-scale board vendors that basically do a lot of taking from the community and don't give anything back, but that's another topic for another day.) I should also mention that I strongly believe that competition (or maybe co-opmetition?) helps keep the world progressing and keeps all the players on their toes. At the same time, I believe there's been more than enough mud-slinging both in this list and in other venues, and we should be able to each make our own decision on the boards based on their technical merits, and not on vague generalities or past mistakes. -Jared Smith I have been in the Asterisk community a little longer than you. I hope you do not consider sharing my experiences as mudslinging. If you do, you should make sure you have your Digium hat and socks off and reconsider. Past experience is one of the things that makes this list great. Anything other than someone's past experience is theory. Here is an idea. Let me exchange my old boards (gathering dust) for your New and Improved boards and I will try them in production. If I do not experience the same issues from the past, I will gladly go back to Digium hardware and sing praises to the lists. If I have the same issues, then that is a different story. The bottom line (and what you conveniently snipped) is that I do what is best for my customers and myself. I live by the cliche (slightly modified for this situation) Sell Me Sub par Hardware Once, Shame on You. Sell Me Sub Par Hardware Twice, Shame on Me I know what works and I generally stick to that. If there has been such vast improvements in hardware, how come Digium is not doing press releases or posting the improvements to the list or a changelog somewhere? Again, take off your Digium hat and socks and think about it. It is a valid question and you raised it yourself and should probably be raised with Mark and the Adtran guys (maybe even the 3com guys too ;-) Again, I hope nobody takes this as mudslinging or bashing Digium, I would be more than happy to buy Digium products if I had the faith that was lost along with many hours of troubleshooting, losing customers, one thing we can all sympathize with, stress. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by
Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released
Just forget it about the 1.2 mantra, it's not going to happen. Focus your energy elsewhere. Lot's of bug fixes are good. Even Cisco comes out with regular bug fixes for IOS. Open source just makes things more visible. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 02, 2007 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released Don Pobanz wrote: the Asterisk release contains a large number of bug fixes for all parts of Asterisk. I am thankful to see the amount of fixes that have gone into this release. However, seeing this many fixes does not give me a warm fuzzy feeling that we won't see a lot more fixes in the near future. So are bug fixes good or bad? ;-) And more importantly, will any of the remaining bugs bite me? Branch 1.4 has one important to us feature that 1.2 does not and that is the queue autofill option. Because of this one feature, I have been wanting to switch to the 1.4 branch for some time. We have a backup system that I will be using for testing. If all goes well, we will move to the 1.4 branch. I hope many others are doing the same so the stability of 1.4 can be improved to the point where no one is concerned. Thanks to all the developers for improving an already great product! Don Pobanz Another reason to call for a 1.2 spoon or fork! Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Supermicro PDSME+ and TE110P
The SME+ has a PCI bridge chip btw the 133MHz and 100Mhz slots so you might want to try moving it to a slot on the other side of the bridge. I believe that card will work in any of the slots. -Original Message- From: kido [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 19, 2007 6:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Supermicro PDSME+ and TE110P Hello all, Has anyone use the Supermicro PDSME+ in combination with the TE110P successfully? My experience so far is not very good. I am running trixbox 2.0 but: 1) with zttool I am getting IRQ Misses. Don't seem to have IRQ conflict, but I am now running my SATA HD in DMA. And I am not able to set it in DMA(HIO... operation not permitted) 2) With zttool the Alarm is RED 3) When I do the loopback cable Pin 1 going to 4 and Pin 2 going to 5, the status led does not turn green... Any Hints? Ideas? Thanks K. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does everyone seem to dislike *now?
You cannot set up your dialplan with the CLI or am I missing something? Creating relatively simple dialplans manually can be quite time consuming. A GUI takes care of all that grunt work. -Original Message- From: SIP [mailto:[EMAIL PROTECTED] Sent: Monday, September 17, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why does everyone seem to dislike *now? Not at all relevant to your query, but I still use the mysql CLI for any mysql task... and while most OSs have nice, functional tools to add users (command-line tools), there are SOME (*cough* Irix *cough*) where there are no CLI tools and VI is your only option (especially if you're remotely logged in via a term window and have no X). GUIs have their place. But it's often a trade-off between abstracting the details to make things user-friendly and hiding the power that is available via the CLI from someone who knows it. If you're comfortable with the CLI, why learn another tool? If you're NOT comfortable with the CLI, by all means use a GUI, but don't expect people who never use it to be of much help when you ask questions. That being said, I like AsteriskNow's GUI. They've obviously spent a lot of work on it (prettier than the stuff that comes with Trixbox). However, for me, I learned using vi and the cli, so I can never quite find what I'm looking for in AsteriskNow. N. Jim Canfield wrote: Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few quesions in #asterisk...my biggest mistake was mentioning *now and I was quickly marked as the GUI idiot. Not entirely untrue at this point but not helpful for someone who is getting started. Here are my first impressions: * The Devs have spent a LOT of time on *now and seem to be doing a fantastic job. * *now is not just a GUI...it's a complete base/reference system - I like that the MOST. * *now is a great starting point for someone new (Me). * *now needs documentation! I know it's in beta, but having links to a down site, is not cool. (Sign me up for help if needed). * *now could be more geared for use as a universal tool. The default contexts and files were quickly replaced with more standard configs. * *now could be very helpful in tracking issues with links to Report a problem or search the WIKI from the app. I understand the tendency to love the CLI, but I honestly think there a place for a GUI in Asterisk. How many of us still use the mysql CLI? I can't expect my helpdesk guy to know emacs or vi just to add a user. Curiously, jc ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
Sorry but your not going to drag me into another one of these you ungrateful bastard type arguments. If you want to take the pepsi challenge as to who contributes how much in what way then email me offline with your list and I'll send you mine! -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Friday, September 14, 2007 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG) shadowym wrote: Yes thank you for reminding me it is open source. Thank you for reminding me people can write their own code for it. I'll get right on rewriting the entire sip code. Should only take me a few hours. Including a couple hours to learn how to write c code. How hard can it be! I can't tell whether you're intending to prove the point that was being made, or trying to be sarcastic. Knowing your posting history, I'll assume the latter. But in case you're serious, and you really do believe the coders owe you something, here's another translation of the situation: If you code, if you contribute to the coding effort by intense testing and/or filing bug reports, if you carry Red Bull to the programmers during hacking sessions, etc., then--in the vernacular of the Church of the Subgenius--you buy slack. And once you have slack, you can say, Let's do this, or Let's do that, and the developers will consider it and--maybe--implement it. When, instead, you are 100% slack-free and have been noted before nipping nasty mots at the hands that feed you code, the chances of having your tart remarks about SLA taken seriously are pretty slim. But, and here's the point: It's Open Source. If the developers look the other way when you ask for something, if they don't answer your emails, if they don't drop everything when you demand something and do what you want, FORK IT! Take the code THEY they wrote and do with it what you will. It's free. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
You never really specified what you want this for. If it is for enterprise type installations then Aastra has a very robust SIP DECT solution specifically designed for multiple roaming extensions. When you go through their webinar training they provide all the calculations in terms of square footage site surveying, noise factors, expected call volume etc. -Original Message- From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] Sent: Friday, September 14, 2007 6:33 PM To: Tilghman Lesher Cc: Asterisk -Users Subject: Re: [asterisk-users] DECT SIP phones On Fri, 2007-09-14 at 12:00 -0500, [EMAIL PROTECTED] wrote: Date: Fri, 14 Sep 2007 09:32:35 -0500 From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] DECT SIP phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. Can anyone suggest a solid alternative DECT SIP phone that is available in North America? I don't know how solid you would consider them, but I have repurposed the ATS X10001P phones that are sold for use with Lingo into phones that can be used with Asterisk. At $70US, I suspect they are the least expensive SIP DECT phones available. Wal-Mart sells the ATS X10001P for $55, and claims it has a fax port: http://www.walmart.com/catalog/product.do?dest=97product_id=6457851 sourceid=1503142050 . Is there a way to fax with these phones without Lingo? How does Lingo do it (over the phone's Internet connection), if Asterisk can't? http://asterisk.drunkcoder.com/hacks/ats-config/ Your server seems very slow, often timing out. -- Tilghman -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
Yes thank you for reminding me it is open source. Thank you for reminding me people can write their own code for it. I'll get right on rewriting the entire sip code. Should only take me a few hours. Including a couple hours to learn how to write c code. How hard can it be! -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Thursday, September 13, 2007 1:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG) shadowym wrote: Maybe his comments were taken out of context as they don't have the whole interview posted. Why is he talking about queue games, Biologicall and other extremely niche crap when there are huge holes in the basic offering (SLA and SCA)? Considering it is an open source project, anybody that has access to the source code (i.e. everybody) can work on whatever they want to, whether it be SLA, SCA, or queue games for the more light hearted. Matthew Fredrickson From: Al lists [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 11, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG) I liked the queue game concept! although it could be cruel! On 9/11/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?
FreePBX is a beautiful thing but nothing will prevent the inevitable train wreck if your hardware is garbage. Not saying yours is but..just sayin. -Original Message- From: Jay R. Ashworth [mailto:[EMAIL PROTECTED] Sent: Thursday, September 13, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly? I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6 stations, mostly IP (I'm looking at the Grandstream 201, to start), and maybe X-lite on a couple of laptops via VPN. We've got a 4xFXO box we bought off eBay, which unfortunately I can't find to quote a model number off of, but I *think* it's a Grandstream as well. I've looked at several of the packages that turn Asterisk from a PBX construction kit into an *actual* PBX, and so far FreePBX looks like the one that matches my mental model of a small phone system best. Anyone have any first hand experiences with it that they'd like to share? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
Maybe his comments were taken out of context as they don't have the whole interview posted. Why is he talking about queue games, Biologicall and other extremely niche crap when there are huge holes in the basic offering (SLA and SCA)? From: Al lists [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 11, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG) I liked the queue game concept! although it could be cruel! On 9/11/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Opinions on AsteriskNOW
Tried the AsteriskNOW beta6 VMWare image yesterday. It's come a long way since last time I looked at it a few months ago. Some things are nicely polished and worked very smoothly but somethings were surprisingly flaky. Maybe because I never had all the incoming/outgoing/user stuff configured as I just wanted to mess around with the GUI. It just seemed to get lost and confused a lot. How does a fully configured setup work? Seems all the important features to make it a viable alternative to some of the other GUI's are there now more or less. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 30, 2007 1:20 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 116 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
Just IMHO but you shouldn't be doing regular updates on a phone system that is working well unless you are doing it to fix a specific problem. It's a phone system not a server. I mean security upgrades as well. At least not until they have been out there for a considerable amount of time. Yea I know Digium says you should upgrade to fix this dangerous security hole immediately sometimes butagain just IMHO..the odds of you having problems with an unpatched system from some new vulnerability are much lower than you messing something up by updating your system. Once it's in production and working well, ultra conservative walking on egg shells second guessing any changes/updates you may be thinking about etc. is the way to go. -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 29, 2007 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] where is 1.4.12? I guess that's my point. I realize asterisk is open source and FREE, however, I wouldn't expect a commercial application to crash as often as I've seen asterisk go down. Don't get me wrong (and we're kind of going way off topic here), I really like asterisk, have done some bug tracing but I don't posses any kind of programming know-how with C... so fixing bugs is out of my court. Again.. asterisk is an amazing product. However, I guess what I'm saying is, I've seen one too many security upgrades take a system down because they induced new bugs. Or a feature upgrade that causes things to be broken (we're talking simple dot upgrades like 1.2.6 to 1.2.7 or something like that). I guess my request is just that Digium maybe spend a little more time in QA before rolling a release out the door. It's just annoying when you do what should be a dot upgrade, and find out a feature that had worked just one dot below has now stopped working, or worse yet asterisk segfaults.And when it's on a production system you can't just keep trying and get traces. On 8/29/07, shadowym [EMAIL PROTECTED] wrote: I have found the response to bug reports extremely impressive! If something happens and I spend a bit of time to get good information to post to bugs.digium.com or put it in a bug thread that matches the problem I am having the response often can be very quick and sometimes resolutions can come with days or even hours. Not just from Digium but 3rd party individuals as well. These are usually not trivial bugs either but often very deep hard to reproduce bugs. I KNOW for a fact if I did have these problems with just about any other commercial product (they all have problems, you just don't know about them until they happen to you) out there I would be SOL or have to put in a lot more effort/time to get things moving forward towards a solution. This is a VERY powerful advantage of Asterisk that should NOT be overlooked IMHO. -Original Message- From: Russell Bryant [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 29, 2007 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] where is 1.4.12? Matt wrote: Just to chime in.. we still have a few systems running 1.2.6 because of Digium's inability to fix bugs. Every version of Asterisk we've ever tried has some sort of major bug that causes it to crash (it being Asterisk) after being up for some period of time, or something doesn't work right... then you'll have version X and version Y will come out as a security fix only, yet stuff is broken in Y that wasn't broken in X. Digium's inability to fix bugs. What a troll ... I'm sure you have never reported any of the issues you have experienced, either. We surely can't fix them if they aren't reported. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
Then you should probably use a commercial application like the Business Edition. I've found that once I decide to go down the open source road it's a different ball game. Test with the latest and greatest release that has the features you need. If it's a fairly new release chances are it's not quite ready for prime time. Open source it not the place to be bleeding or even leading edge and expect a smooth ride. -Original Message- From: Jay R. Ashworth [mailto:[EMAIL PROTECTED] Sent: Thursday, August 30, 2007 5:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] where is 1.4.12? On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote: I guess that's my point. I realize asterisk is open source and FREE, however, I wouldn't expect a commercial application to crash as often as I've seen asterisk go down. Windows 98. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
It depends. I won't upgrade to the latest released Cisco IOS either unless I need the added features and only after it's been out for several months. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Thursday, August 30, 2007 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] where is 1.4.12? shadowym wrote: Then you should probably use a commercial application like the Business Edition. I've found that once I decide to go down the open source road it's a different ball game. Test with the latest and greatest release that has the features you need. If it's a fairly new release chances are it's not quite ready for prime time. Open source it not the place to be bleeding or even leading edge and expect a smooth ride. And closed source is? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where is 1.4.12?
I have found the response to bug reports extremely impressive! If something happens and I spend a bit of time to get good information to post to bugs.digium.com or put it in a bug thread that matches the problem I am having the response often can be very quick and sometimes resolutions can come with days or even hours. Not just from Digium but 3rd party individuals as well. These are usually not trivial bugs either but often very deep hard to reproduce bugs. I KNOW for a fact if I did have these problems with just about any other commercial product (they all have problems, you just don't know about them until they happen to you) out there I would be SOL or have to put in a lot more effort/time to get things moving forward towards a solution. This is a VERY powerful advantage of Asterisk that should NOT be overlooked IMHO. -Original Message- From: Russell Bryant [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 29, 2007 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] where is 1.4.12? Matt wrote: Just to chime in.. we still have a few systems running 1.2.6 because of Digium's inability to fix bugs. Every version of Asterisk we've ever tried has some sort of major bug that causes it to crash (it being Asterisk) after being up for some period of time, or something doesn't work right... then you'll have version X and version Y will come out as a security fix only, yet stuff is broken in Y that wasn't broken in X. Digium's inability to fix bugs. What a troll ... I'm sure you have never reported any of the issues you have experienced, either. We surely can't fix them if they aren't reported. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server recommentation (unique requirements)
We are at about 250 days of 24/7 uptime now. It would be more but we had a long power outage and the UPS's ran out. We are using Sangoma cards though. You can easily substitute the 2U for a 3U but I don't think you need it. Qty 1 Supermicro SC823T-R500LP, 2U, redundant 500W ps w/ PFC, 6x1 SATA hot swap bays, DVD-RW/Floppy http://www.supermicro.com/products/chassis/2U/823/SC823T-R500LP.cfm Qty 1 Supermicro PDSME+, Intel 3100 Mukilteo-2p chipset, Dual LAN, 2x 64-bit 133MHz PCI-X, 2x 64-bit 100MHz PCI-X, optional KVMoIP card http://www.supermicro.com/products/motherboard/Xeon3000/3010/PDSME+.cfm -Original Message- From: Stephen Kratzer [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 28, 2007 7:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] server recommentation (unique requirements) Howdy. I've been having trouble finding a fairly modern server that meets the following requirements: - Molex power connectors (don't want to use the Digium FXS power supply) - 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC) - dual power supplies - preferably dual CPUs = 1GHz - preferably rack-mountable (3-4RU) - CentOS-friendly We'd also like to stay away from older HP servers. Any recommendations would be greatly appreciated. Thanks. Stephen Kratzer Network Engineer CTI Networks, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
They know what they are doing and do a lot of it. I don't have to give an opinion myself. There is enough evidence all over for people to draw the proper conclusions for themselves. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma On 8/26/07, shadowym [EMAIL PROTECTED] wrote: Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. Can you define 'serious'? -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
And the FUD continues. Pain and misery eh? Google pain misery insertmodel#here? -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma shadowym wrote: Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. When did saying nothing at all become enough? Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. Yep. Right now they can't work with the 2.6.22 kernel. I know annoying kernel changes cause them trouble, but they don't respond with the speed they should. Most people want to keep their platforms fully updated, and for many that means the 2.6.22 kernel is going onto their system around now. They also keep poor notes. For example, when 3.1.0 became necessary to be able to use a recent kernel, finding out that is was necessary took some effort. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. I've never had a panic, but I have no expectation of a smooth ride when updating the Sangoma drivers. Pain and misery is a more the norm. It is best to wipe out everything you can find on your machine related to wanpipe and zaptel before an upgrade. They seems to end up using bits of old material under some circumstances, causing strange results. Regards, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
Still true on CentOS 5. You can only RAID partitions unless you do the LVM thing. What are the disadvantages compared to being able to RAID the whole disk? Maybe for monitoring it's just more to deal with but does it make a RAID 1 any less reliable? -Original Message- From: Zane C.B. [mailto:[EMAIL PROTECTED] Sent: Thursday, August 23, 2007 9:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk On Wed, 22 Aug 2007 12:37:26 -0600 Stephen Bosch [EMAIL PROTECTED] wrote: Zane C.B. wrote: 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. How is this any different in FreeBSD? Could you explain to me how else you are going to mirror an entire disk in software when your boot partition is on the disk? The raid info is done the same as on other decent system, it is stored at the in the last sector of the provider. making a mirrored freebsd system is like this... 1: install freebsd 2: dd if=current drive of=2nd drive for mirror 3: gmirror label some name 2nd drive 4: mount 2nd drive and edit fstab to boot using /dev/gmirror/whatever 5: boot from 2nd drive 6: gmirror insert name original drive /me loves GEOM, the goddess of all disk subsystems or whatever. http://www.freebsd.org/cgi/man.cgi?query=gmirrorapropos=0sektion=0manpath =FreeBSD+6.2-RELEASEformat=html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem compiling Zaptel 1.4.5.1
It compiles fine for me but I can't change the soft EC. It always compiles with MG1 no matter what I select in zconfig.h. Downgraded back to 1.4.4 and it works fine again. -Original Message- From: Jan du Toit [mailto:[EMAIL PROTECTED] Sent: Friday, August 24, 2007 1:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem compiling Zaptel 1.4.5.1 Hi. Please help. When trying to compile Zaptel 1.4.5.1 I get the following: /build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c base.c:48:29: linux/workqueue.h: No such file or directory base.c:292: warning: `vpm150m_firmware' defined but not used make[2]: *** [base.o] Error 1 make[2]: Leaving directory `/usr/src/zaptel-1.4.5.1/wctdm24xxp' make[1]: *** [wctdm24xxp/wctdm24xxp.o] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.5.1' make: *** [all] Error 2 Can anybody help me with this? I run make distclean, configure and then make. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
I've gotten burned by software raid so I'll probably be sticking with hardware in the future. If your drive dies for some reason it could affect the SATA bus and cause the system to crash. That's what happened to me. It wouldn't come back up on the second Raid 1 drive until I removed the bad one. With hardware raid that would NEVER happen. They are designed to isolate the Sata interface of a drive gone bad and just keep on running. Are there Hardware Raid cards that don't come with software for monitoring and emailing warnings? I know the 3ware cards come with linux software that can run as a web interface or command line so the whole monitoring argument is mute IMHO. -Original Message- From: Arnaud Ligot [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 21, 2007 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk My servers run in a datacenter, 50km away from my office... if a led flash, if the speaker beep... I think I'll not see/hear it ... My servers are monitored using nagios which has a plugin for software raid... so if one array goes down, I receive a mail/sms/call/... futher more, everything is on the same panel: raid, http servers, free disk space, ... I think it is better than any led flashing into the DC :-D A. On Tue, 2007-08-21 at 10:30 -0400, Steve Totaro wrote: I thought that was what the flashing LEDs on the front of the server's HDs were for (besides showing activity). Some I have seen also have an LED near the power button to indicate HD problems. I guess if you are building your own boxen and not using enterprise grade servers, this is not the case. Thanks, Steve Totaro C F wrote: While hardware RAID tend to be more reliable, it is not always possible to properly monitor hardware raid in a linux system, unless you write your own code. Consider this: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Try some of these suggestions. http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 15, 2007 12:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM400P FXO click sounds Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. [channels] language=es context=ent-4229 ;rxwink=300 usecallerid=yes hidecallerid=no ; Whether or not to enable call waiting on internal extensions ; With this set to 'yes', busy extensions will hear the call-waiting ; tone, and can use hook-flash to switch between callers. The Dial() ; app will not return the BUSY result for extensions. ; callwaiting=yes threewaycalling=yes transfer=yes ;canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=128 relaxdtmf=yes rxgain=3.0 txgain=3.0 callgroup=1 pickupgroup=1 immediate=no ;busydetect=yes ;busycount=4 callprogress=no ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ;callprogress=yes faxdetect=incoming faxdetect=outgoing signalling=fxs_ks group=1 channel=1 signalling=fxs_ks group=2 channel=2; singalling=fxs_ks group=3 channel=3; ;singalling=fxs_ks ;group=1 ;channel=4 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Please explain to me how FXO tune would fix popping and clicking sounds??? -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 15, 2007 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400P FXO click sounds [EMAIL PROTECTED] wrote: Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the same jacks, and the line is crystal-clear. Below is my zapata.conf, if you guys have any ideas how I might resolve this, I'd appreciate it. I have installed from sources Asterisk 1.2.22 and zaptel-1.2.19 on a debian etch x86_64. Have you tried running fxotune on it? If you have (or haven't) make sure you try the zaptel-1.4 version of fxotune. It has improved significantly since 1.2. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
Unless it's Monster cable ground wire at $100 a foot that has depleted oxygen and is bombarded by Xrays to free up the quantum particles which makes everything work better. [/sarcasm off] -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Thursday, August 09, 2007 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major Digium Card Problems Jay R. Ashworth wrote: On Wed, Aug 08, 2007 at 11:44:51PM -0400, Michael J. Liberatore wrote: First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. First, find the knob in your mailer that says send messages as HTML and turn it off, please? HTML is bad for mailing lists. Secondly, remember: this is a *phone* system now; you're hooking it up to several kilofeet of antenna. If you don't have telco-quality lightning protection and grounding on the box, you can expect this sort of thing. You can't find practices handbooks anymore (damnitall), but if you've ever looked at a professionally installed key system backboard, and seen those Porta-Systems gas-tubes, and the size of the grounding wire, then you may get an inkling of a) why you're having problems, and b) why traditional PBX's cost so much to buy and install. It's not *all* extra markup, folks. Cheers, -- jr 'hobby horse' a I was not aware that ground wire was very expensive or difficult to ground correctly. I do not see how that adds very much to the dealer's cost. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] v1.4.x ready yet?
Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hot GXP-2000
Yikes!
{sarcasm on}
Yea, why use good stuff when you can get stuff at less than half the price.
Who cares if it ACTUALLY works properly. As long as it's cheap.
{/sarcasm off}
Yikes!
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 09, 2007 10:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hot GXP-2000
On 6/10/07, Bill Hackensack [EMAIL PROTECTED] wrote:
Why? Because of their excellent customer support in taking care of a
problem? At the price of the Grandstreams compared to others, I can
deal with a couple of bad apples. I can buy two Grandstream's for the
price of a phone with similar features. I can deal with a lot of bad
apples at that ratio. Their sidecar is so cheap compared to others it's
not even funny.
Plus, I can't even get some of the functionality the GXP's give me
from other phones.
However the problem is it's not even a phone.
On 6/9/07, Dovid B [EMAIL PROTECTED] wrote:
One of the reasons why I stand clear of Grandstream
- Original Message -
From: Carlos Chavez [EMAIL PROTECTED]
To: Asterisk asterisk-users@lists.digium.com
Sent: Friday, June 08, 2007 6:47 PM
Subject: [asterisk-users] Hot GXP-2000
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RE: [asterisk-users] Noise on FXS ports (Sangoma)
All I know is that your rx gain, at 0db is probably way too low. My tx gain is typically set to -3db. Yes, I was thinking sip phones when I made the amplifier comment. If you hear the noise as soon as you take the analog phone off hook without dialing the PSTN then I don't see how it could have anything to do with the telco card. Is it a feedback type noise or background hiss? Feedback type buzz would lead me to believe it's ground loop as someone else mentioned. Ground loop hum is a PITA to try deal with. Yes, ztmonitor works with Sangoma cards. It's fxotune that doesn't work and is not needed on Sangoma cards. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 05, 2007 11:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Noise on FXS ports (Sangoma) shadowym wrote: Every Sangoma A200 card I have ever connected to the PSTN required a rx gain of at least 10. Yours is commented out which I believe would make it default to 0? The noise is present on FXS ports only and is audible the moment the receiver is lifted. I mention this because you say Every Sangoma A200 card I have ever connected to the PSTN. The affected ports aren't connected to the PSTN. I am guessing that because the rx gain is so low the users are cranking up the phone volume all the way and maybe your hearing amplifier background noise?? Well, these are simple, analog sets. There's no amplifier or gain control on the set. They have the traditional varistor for amplitude balancing. If what you say is true, the varistor might be compensating so aggressively that it's making the noise audible. This explanation seems inadequate, though, as I don't hear this sound on any of the SIP phones; only the analog phones connected to FXS ports. A Sangoma engineer told me to increase the txgain, but as I mentioned, that didn't help. Your should run ztmonitor and adjust your gains. I didn't know it was appropriate to use ztmonitor with Sangoma hardware. -Stephen- ___ --Bandwidth 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] Noise on FXS ports (Sangoma)
I don't see any mention of you adjusting gains on the card/phones. Also, what are you doing for echo cancellation? Can you post your zapata.conf file? -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Monday, June 04, 2007 10:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Noise on FXS ports (Sangoma) Stephen Bosch wrote: Stephen Bosch wrote: Hi: I have a Sangoma A200 card installed in a server with two FXO modules and one FXS module. Analog sets connected to the FXS module have a squeaky static -- it's like static mixed with the sound of someone vigorously cleaning a window a few doors down. In other words, it's not a classic static noise, but it is noise, and it's distracting. Remote callers can hear this noise. I have, by turns: - Tested the line with the same analog sets plugged in at the demarcation point. No noise. (None of the SIP devices in this configuration have this noise for outgoing calls, so I'm sure it's got something to do with the FXS module). - Plugged the known good analog sets straight into the server - Moved the FXS module off the REMORA daughtercard to the main card - Replaced the FXS module with a new one - Run the server off of battery power, to see if the noise is garbage leaking in off the AC - Turned off the PoE midspan The next thing I'm going to try is turning off the switch it's plugged into. When that is done, I'll have done everything short of move the server to a different location. Maybe the power supply is generating this crap. Hmn. I'm going to test this hypothesis. Okay -- that didn't work. I swapped the power supply out with a better one, and even disconnected the extra ventilation fan. The noise is different but still there. To those with experience with FXS modules, I welcome your input. -Stephen- ___ --Bandwidth 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] Noise on FXS ports (Sangoma)
Every Sangoma A200 card I have ever connected to the PSTN required a rx gain of at least 10. Yours is commented out which I believe would make it default to 0? I am guessing that because the rx gain is so low the users are cranking up the phone volume all the way and maybe your hearing amplifier background noise?? Your should run ztmonitor and adjust your gains. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 05, 2007 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Noise on FXS ports (Sangoma) shadowym wrote: I don't see any mention of you adjusting gains on the card/phones. Also, what are you doing for echo cancellation? Can you post your zapata.conf file? I had actually tried to adjust the gains, but it actually seemed to make the problem worse. I turned echo cancellation off on bridged calls because it did not seem to be necessary and did weird things to the sidetone Here is the zapata.conf: ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes ;callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes ;echocancelwhenbridged=yes ;rxgain=2.5 ;txgain=4.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:8 bus:1 span:1] context=private-incoming group=0 signalling = fxs_ks channel = 4 context=office-incoming group=1 signalling = fxs_ks channel = 3 context=office-incoming group=1 signalling = fxs_ks channel = 7 context=fax-incoming group=0 signalling = fxs_ks channel = 8 context=fax-internal relaxdtmf=yes group=2 signalling = fxo_ks channel = 2 context=private-internal relaxdtmf=yes group=2 signalling = fxo_ks channel = 1 -Stephen- ___ --Bandwidth 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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
Can you cut and paste the last few relevant lines of your log file? That should help determine what is causing the core dumps. After that is determined you can file a bug report with the log file cut and paste if necessary. Is there some reason you cannot test patches on a separate test system. If it's a legitimate bug there will likely be others looking for a solution that would be willing to test. Doing things that way helps everyone including yourself. I have found the developers VERY responsive to well documented bug reports. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 29, 2007 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW * Tzafrir Cohen wrote: On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote: Michael Collins wrote: I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . Fair indeed. I would guess that a completely stable 1.2 w/ security maintenance is acceptable to the majority of users. Those folks still using 1.0.x certainly aren't clamoring for new features! The great many Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a former consultant for these customers. So basically what you're saying is that some efforts should be concentrated on 1.2 as well. So let's start with your specific problems. Are there open bugs for them? My specific problem is that Asterisk 1.2.17 and 1.2.18 (I've not tried 1.2.16) core dumps at least once per day. 1.2.15 works just fine for me. I don't know if there are open bugs. I've not opened any bugs. Any time I open a bug for a problem I have on a production server, all people want me to do is test patches to see if they fix the issue. They don't seem to understand the term production server. Sorry, but this is my JOB, not my hobby. Perhaps the customers of the developers don't care if the PBX crashes once per day, but my customers do care and I will do whatever is required to make them stop yelling at me. What made them stop yelling at me is moving back to 1.2.15. ___ --Bandwidth 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] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
If anything this should motivate the FreePBX developers a bit more. -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 30, 2007 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO! On 5/30/07, BSumrall [EMAIL PROTECTED] wrote: AMP does not support 1.4 and will not until AMP 2.3 is released! I'm sorry to hear you think our decision (I say our, as I was at the Asterisk Developers' Conference where the decision was made) will kill the AMP project. Personally, I don't think the situation is as dire as you say. I'm quite sure the AMP developers will step up to the plate and support Asterisk 1.4 in due time. When that will be I can't say, as I'm not active in the AMP community. I can't image it would take that long to move over to Asterisk 1.4, as the dialplan changes aren't *that* extensive between 1.2 and 1.4. (Obviously any code that ties into the internal C APIs of Asterisk will take longer to port.) Bet you guys didn't think about that one! Actually, we did. As a matter of fact, I was *very* vocal at the conference in stating that we needed to give users, integrators, and projects like AMP a substantial warning before putting Asterisk 1.2 in security maintenance mode, as they need time to react. At the same time, I don't think anyone should expect the Asterisk developers to base all their decisions completely on the timetables of outside projects (like AMP). There is a plethora of projects and programs out there that tie into Asterisk, and if we as developers waited for every single one to move over to Asterisk 1.4, we'd never accomplish anything. There's simply a finite set of resources (developers and bug marshalls in this case), and a decision had to be made on how best to use those resources. Personally, I think it would be great if there were more communication between the outside projects and the Asterisk developers, so that there isn't so much animosity when decisions like this are made. In short, the decision is probably going to cause some short-term discomfort for some people, but I truly believe it's a good decision for the long-term health and sanity of the Asterisk developers and Asterisk community in general. No, we're not trying to kill off AMP or any other outside project -- we're trying to make Asterisk (and by extension, anything that uses or adds on to Asterisk) as great as possible. -Jared ___ --Bandwidth 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] Bottom line on fax reception
Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Octasic echo cancellation
Hi Sebastien, I'm just a lowly user but I will tell you what I think I understand about it. There is nothing in the Octasic documentation that suggests you can have continuosly updated statistics but I agree that would be a nice to have feature. Have you tried contacting Octasic about that? You can set the ERL and tail length in the octveqd.conf file which you need to manually copy into the /etc/ directory as per the Octasic documentation and octveqd.conf file included with softecho. By default they are set to 9db and 64ms respectively. It is my understanding EC takes place AFTER the tx/rx gain as part of the Zaptel driver function. Therefore your tx/rx gain WILL have an effect on it's operation and (I think) the ERL values. If your using Digium cards you should run fxotune and set your gains. With Sangoma cards you only need to set your gains. If that is all set up ok softecho should get rid of all your echo. If not I would suggest you try increase the tail length to 128ms in octveqd.conf. I don't think you should need to change any of the other settings. -Original Message- From: Sebastien Leclere [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 10:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Octasic echo cancellation Hi, I'm currently testing SoftEcho, an echo cancellation software for Asterisk from Octasic. I noticed an important increase of the quality of my coms, but I still have a few echo problems. There is an ERL parameter which corresponds to an initial ERL value probably to optimize the echo training or something in that kind. Is there a way to monitor, using one of the zttools, the instant ERL value, to be able to set this parameter correctly? In that purpose, I would like to know where do the echo cancellation take place in the communication chain, is it after or before the amplification by the Rx/Tx gain parameters? Will this gain apply on the echo cancelled signal or on the gross signal? Anybody with experience on this product could give me some advice in the aim of removing all trace of echo? Thanks by advance Sebastien ___ --Bandwidth 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] Bottom line on fax reception
Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bottom line on fax reception
Thanks for all the replies. Seems there are at least 2 or 3 people giving strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade production) solution. That is just the sort of feedback I was looking for. My application is just standard business reception of faxes. Right now they use WinFax and probably receive about 30 to 50 faxes a day. I want to wean them off Winfax as it's not really supported anymore and I dislike all things Symantec in general. Receiving faxes on Asterisk has the added benefit of being able to use the fax line as an extra outgoing line when the rest are in use. That is what they are doing now on their key system and they don't want to lose that ability. I don't blame them. -Original Message- From: Duncan Turnbull [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no baby sitting, I receive about 20 and it requires no baby sitting Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and hylafax lists for much bigger examples Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 29 May 2007 7:34 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conference room as Music on Hold
Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radio station. I assigned mono out in XP Equalizer as the mic so now I have the softphone streaming audio. I then create a conference room and dial that room from the softphone. Now anyone who joins the conference room hears the streaming audio. How can I configure Asterisk so that when musiconhold is invoked it automatically joins that conference room? The reason I want to do this is because the radio station uses windows media and I haven't been able to get it to work directly on Asterisk. Even after following the mplayer instructions on the Asterisk wiki. ___ --Bandwidth 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] Windows Media streaming for MOH?
Anyone have Windows Media streaming for MOH working? I followed the various procedures on the Asterisk Wiki for using mplayer which seems to be the only Linux player capable of playing windows media streaming audio (asf, wmv etc.). Anyone get this working? I can get shoutcast streams working using mpg123 but so far no luck with windows media streaming. Is there another player out there or a trick of some sort? I've been googling but so far no luck. The problem is that many radio stations including the local ones people in my area use for MOH on their traditional PBX's use windows media streaming. As a work around I would consider streaming it from a softphone on a Windows PC to a conference room if I had to. That may be easier to do but haven't found much info on that either. Worst case I would consider a receiver on Line In on a softphone on a Windows PC or absolute worst case, from line in audio on the Asterisk server. Streaming local radio directly on the Asterisk server is the most elegant solution IMHO. Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Aastra MWI
This is probably cold comfort but I have NEVER had any issues with MWI working on Aastra phones. It always just works by default. No extra configuration necessary on the phone for sure. Just reset it to factory defaults. Explicit MWI is NOT checked by default and I have never had to check it. No extra configuration on Freepbx/Trixbox. Not sure about a basic Asterisk install but here is my sip.conf. [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying The number you have dialed is not in service. Please check the ; number and try again. context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf [600] type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=no [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all dial=SIP/600 context=from-internal canreinvite=no callerid=device 600 allow=ulaw -Original Message- From: Lee Jenkins [mailto:[EMAIL PROTECTED] Sent: Monday, May 21, 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Aastra MWI I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting Explicit MWI Subscription to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, 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
RE: [asterisk-users] HPEC audio clipping
Octasic SoftEcho works very well for me. _ From: Olivier [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 6:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HPEC audio clipping 2007/5/15, George Pajari [EMAIL PROTECTED]: If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. Note, however, that we have observed stability issues with HPEC 8.2 (two kernel panics in two weeks since installing HPEC). (The 9.00 version had such severe clipping that we could not run it long enough to determine if it was more stable). g. So, what's going on ? Should someone file a bug report, so that we can check progress on this ? We have customers waiting for echo cancellation improvement. As we already tried to use HPEC, it would be very hard to either try something different nor to wait without deadline. So ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: RE: Digital Phones
You can use the digital phones with Asterisk. Just that you need a Citel Portico external box to convert the proprietary (ie. Digital) non IP protocol that cannot work on TCP/IP networks to standard IP SIP protocol that can. At about $120 per port the advantages and potential issues may or not make it feasable. If there is already a significant investment in high end digital phones that can cost many hundreds of dollars each and you want to continue to use the existing wiring it is probably worth considering IMHO. My 2 cents. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Thursday, May 10, 2007 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: RE: Digital Phones Tom Lynn wrote: Bilal, I don't think anyone is telling you that digital phones don't need cards. I do think they are telling you that NOBODY makes a card that drives digital phones for use with Asterisk. bilal ghayyad wrote: Hi List; As I know from AVAYA (I am AVAYA certified) that digital phones are connected to digital cards and it does not go through ethernet switches at all, digital phones should be independent on the ethernet network, so if the network down, these phones will start working, it will be totally isolated from the data traffic. So, how that come the digital phones does not need a card for it? It is *not* that the digital phones do not need a card. It is that there is no card on the market that will work with them. Also, how it will use ethernet switches! It will not use ethernet switches. It does not work with IP Packets. No, it does not, and it will probably never work. To make it even clearer: Bilal -- You cannot use your digital phones with Asterisk. Period. Either: 1 - use a matching digital switch (Nortel Meridian or Norstar, etc.) 2 - sell the phones and replace them with proper IP phones that you can use over an IP network. If you are concerned about availability in the event of a power failure: Purchase an Uninterruptible Power Supply of the appropriate size, Power over Ethernet injectors, and configure your system such that it will stay up during a power outage. -Stephen- ___ --Bandwidth 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] SLA broken in 1.4.3?
So how well does SLA work? The FreePBX developers (well, at least one) seem to think it's a non-starter and dumbs down the other features too much. I haven't experimented with it yet so I'm eager to hear some real world feedback. -Original Message- From: David W. Rice [mailto:[EMAIL PROTECTED] Sent: Saturday, May 05, 2007 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SLA broken in 1.4.3? SLA requires meetme which requires at a minimum ztdummy. So, you must compile and install zaptel, then compile and install asterisk 1.4.3 and the sla commands will be in the CLI. Let me know if you need help setting up SLA on Polycom phones with *. I've done it successfully and have the configs. I'm going to put them on the web some time soon. -Original Message- From: Jay Austad [mailto:[EMAIL PROTECTED] Sent: Friday, May 04, 2007 8:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SLA broken in 1.4.3? I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console sla commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 on CentOS 5?
Just wondering if anyone has tried using Asterisk 1.2 on CentOS 5. Is it worth considering for a Production install yet? Did they fix that spinlock.h Kernel problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk locked up
SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could tell. Seems to be running fine now. Here is the log file. I deleted the stuff in the middle to keep it brief. The very last entry is where it locked up and stopped logging/responding. Anyone know what this might be? May 2 11:38:02 VERBOSE[10853] logger.c: -- Playing 'vm-press' (language 'en') May 2 11:38:02 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:02 DEBUG[10853] app.c: Locked path '/var/spool/asterisk/voicemail/default/660/INBOX' May 2 11:38:02 DEBUG[10853] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/660/INBOX' May 2 11:38:02 DEBUG[10853] app.c: Locked path '/var/spool/asterisk/voicemail/default/660/Cust3' May 2 11:38:02 DEBUG[10853] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/660/Cust3' May 2 11:38:02 DEBUG[10853] app.c: Locked path '/var/spool/asterisk/voicemail/default/660/Cust3' May 2 11:38:02 DEBUG[10853] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/660/Cust3' May 2 11:38:02 DEBUG[10853] channel.c: Scheduling timer at 160 sample intervals May 2 11:38:02 VERBOSE[10853] logger.c: -- Playing 'vm-Cust3' (language 'en') May 2 11:38:12 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:12 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:12 VERBOSE[10853] logger.c: == Parsing '/var/spool/asterisk/voicemail/default/660/Cust3/msg.txt': May 2 11:38:12 VERBOSE[10853] logger.c: == Parsing '/var/spool/asterisk/voicemail/default/660/Cust3/msg.txt': Found May 2 11:38:12 DEBUG[10853] app_voicemail.c: VM-Duration: duration is: 48 seconds converted to: 0 minutes May 2 11:38:12 DEBUG[10853] channel.c: Scheduling timer at 160 sample intervals May 2 11:38:14 VERBOSE[10853] logger.c: -- Playing 'vm-deleted' (language 'en') May 2 11:38:14 DEBUG[3778] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 9948: Match Found May 2 11:38:14 DEBUG[3778] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 9820: Match Found May 2 11:38:16 VERBOSE[10853] logger.c: -- Playing 'vm-message' (language 'en') May 2 11:38:17 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:17 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:17 VERBOSE[10853] logger.c: == Parsing '/var/spool/asterisk/voicemail/default/660/Cust3/msg0001.txt': May 2 11:38:17 VERBOSE[10853] logger.c: == Parsing '/var/spool/asterisk/voicemail/default/660/Cust3/msg0001.txt': Found May 2 11:38:17 DEBUG[10853] app_voicemail.c: VM-Duration: duration is: 27 seconds converted to: 0 minutes May 2 11:38:17 DEBUG[10853] channel.c: Scheduling timer at 160 sample intervals May 2 11:38:17 VERBOSE[10853] logger.c: -- Playing '/var/spool/asterisk/voicemail/default/660/Cust3/msg0001' (language 'en') May 2 11:38:24 VERBOSE[10853] logger.c: -- Playing 'vm-goodbye' (language 'en') May 2 11:38:25 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:25 DEBUG[10853] channel.c: Scheduling timer at 0 sample intervals May 2 11:38:25 DEBUG[10853] app.c: Locked path '/var/spool/asterisk/voicemail/default/660/Cust3' May 2 11:38:25 DEBUG[10853] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/660/Cust3' ___ --Bandwidth 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] Digital Phones
You can use something like this which supposedly works well and is easy to configure. Costs about $120 per port full retail and works with all sort of phones including Nortel. http://www.citel.com/Products/Portico.asp -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 01, 2007 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Digital Phones What brand of digital phones, I think I read some time ago that someone was doing something with Nortel phones but I seem to remember the cost of the phone meant...better to toss the handsets and buy new sip handset. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Tuesday, 1 May 2007 6:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digital Phones Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Pix firewalls
Yes, we found (at least with Aastra phones) that we had to disable the SIP fixup protocols on a pix 501. Here is the whole setup. NOTE: I could be wrong but I believe the requirement to open ports 1-2 for remote extensions has become an urban myth. I don't think you need to open any more than you have extensions + maybe a few more as a buffer. That is what we are doing and haven't had any problems. I'm no expert so someone please correct me if I'm wrong. Here is the procedure: start Firewall/Router configuration: The following ports needed to be forwarded to the asterisk server for various remote access Port 80 (Freepbx web access) Port 4445 (Flash Operator Panel web access) Port 4569 (IAX remote phone clients) Port 5059-5061 (registration and proxy server access, default is 5060) Port 1-10025 (ports reserved for RTP voice packets for SIP phone conversations) Aastra Phones as external extensions This assumes the Asterisk server is configured for external extensions and the extension configuration in asterisk is configured to be used as an external extension. Both are described earlier in this guide(sip_nat.conf, nat=yes). Reset the phone to factory defaults. All you need to configure in the phone are phone number, callerID, authentication name, password, Proxy IP and Registrar IP. Leave everything else at default and it should work. I also changed registration retry timer and BLF subscription period to 120s. Special note about Cisco PIX firewall In order to make Aastra phones work outside a Cisco PIX firewall to the Asterisk server inside the firewall, we needed to remove fixup protocol sip 5060, and fixup protocol sip udp 5060 which are both enabled by default. no fixup protocol sip 5060 no fixup protocol sip udp 5060 Special note about extensions over VPN In order to make extensions work over VPN's we had to add the VPN subnets to sip_nat.conf to make the phones on the 192.168.2.0 and 192.168.3.0 subnets work with the Asterisk Server on the 192.168.1.0 subnet. Here is the whole sip_nat.conf file nat=yes externip=xxx.xxx.xxx.xxx localnet=192.168.1.0/255.255.255.0 localnet=192.168.2.0/255.255.255.0 # VPN1 to 192.168.1.0 localnet=192.168.3.0/255.255.255.0 # VPN2 to 192.168.1.0 externrefresh=10 the end -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 24, 2007 8:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Pix firewalls AFAIK asterisk does't yet suport TCP, it is therefore not necessary to open TCP 5060. On Cisco PIX you might also need to disalbe SIP fixup On 4/24/07, Lee Jenkins [EMAIL PROTECTED] wrote: Noah Miller wrote: Hi Don I asked this last week but i didn't get any answer So i will elaborate on my question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip traffic thru it from a sip phone wherever i may be. The pix is where all my servers are colocated and i will need to connect thru it from softphones / hardphones wherever i happen to be traveling. I need help setting up the pix for inbound and outbound sip/iax traffic. Any help would be greatly appreciated. If you're looking for which ports to open: SIP: TCP and UDP port 5060 (signalling) - can be changed in sip.conf UDP ports 1-2 (RTP stream) - can be changed in rtp.conf IAX: UDP Port 4569 Is it possible to reduce the number of ports to be opened if there is moderate traffic? -- Warm Regards, 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 ___ --Bandwidth 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 Pix firewalls
Again, is the 1-2 not an urban myth? Someone correct me if I'm wrong. I run about 10 external extensions and limit the ports to 1-10025. I just can't see why you would need to open 1 ports to the outside world unless your going to have 1 simultaneous conversations. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 24, 2007 9:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Pix firewalls On Tue, Apr 24, 2007 at 11:04:53PM -0400, Lee Jenkins wrote: Noah Miller wrote: SIP: TCP and UDP port 5060 (signalling) - can be changed in sip.conf UDP ports 1-2 (RTP stream) - can be changed in rtp.conf Yes. See rtp.conf (at least on your side). Also, if the firewall understands SIP, it may be smart enough to open the ports for the relevant RTP ports upon the beginning of a SIP session. So consider trying not to open any port for RTP. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Marketing 101
Thanks for the advice. Maybe I should clarify what I was asking. It's not so much the how but the what. What are people doing to get PBX Sales/Support business. I know how to get IT business but potential customers still see the Telco business as quite different and are used to using separate companies for that. What I was asking is how the traditional telco guys get new sales/support/consulting business. With IT it's usually a combination of cold call/networking/word of mouth. I'm hoping that Telco is the same but I never see any telco guys at networking events so I am thinking they cold call and advertise targeted at business owners. I'm not sure though. -Original Message- From: dave cantera [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 24, 2007 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Marketing 101 shadowym, best thing to do is talk to a lot of consultants, coaches, and marketing people... take the approach you do with learning open source only reverse it... instead of reading source (internal) ask people (external)... it is a big undertaking and the most important task you have... marketing is a bigger task than the technical (for a tech anyway) don't go it alone nothing happens without marketing (and sales)... marketing is *not* sales... daveC shadowym wrote: I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth 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 Business Edition Question
If you have an interest in learning a bit of Linux I would suggest looking at Trixbox. I would not have said that 1 year ago but it has come a long ways since then. Eventually as you learn more you can install your own Linux/Asterisk/FreePBX from scratch just for the sake of being able to learn more and have more control and less unnecessary bloat. Trixbox seems to be working fine for most people in production install these days though. There is the option to install AsteriskNow as part of the new TrixBox 2.2 release which is in final beta set to be production released very soon. Asterisk 1.4 and AsteriskNow are still immature so I would not consider either of them for a production install yet. AsteriskNOW is not nearly as full featured as FreePBX yet. They included it as an optional install in the new Trixbox for testing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 25, 2007 2:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Business Edition Question Hi, Can anyone in the list help me with these queries on Asterisk Business Edition. 1. Why would anyone choose the Business Editon when the whole thing is avalable as GPL? 2. Is there a GUI to manage asterisk? 3. Can it be compared with Asterisk NOW? 4. Is the CD a complete installation package? 5. If im looking for hiring a server on a remote location how will i be able to install it? If someone can guide me on it, that would be great. Also i would like the users to share a bit on their experiences with it. Thanks Danny. ___ --Bandwidth 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 Pix firewalls
That is probably because you did not disable SIP fixup protocol. When you set up a PiX correctly it works. Guaranteed! -Original Message- From: Ed Nuñez [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 25, 2007 6:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk Pix firewalls Don This may not be a solution to your question, but I would like to share that Ive been having one way audio issues when connecting point to sight to a PIX 515E using SIP. I changed to IAX and this is working perfectly now. It was paynless to configure IAX2, so you might want to consider it. Ed From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don E. Wisdom Sent: Tuesday, April 24, 2007 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Pix firewalls Hi, I asked this last week but i didn't get any answer So i will elaborate on my question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip traffic thru it from a sip phone wherever i may be. The pix is where all my servers are colocated and i will need to connect thru it from softphones / hardphones wherever i happen to be traveling. I need help setting up the pix for inbound and outbound sip/iax traffic. Any help would be greatly appreciated. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
