[Asterisk-Users] Polycom IP 300 VoiceMail Retrieval
Hello, I have just picked up a pair of SoundPoint IP 300 phones for testing purposes, and they work great. Really good quality units for the price. I only have two complaints/issues with them; 1. The $40 additional cable for POE to work. That sucks. 2. I can't for the life of me figure out the "Single Keypress Voicemail Access" that they are talking about in the manual; According to the Polycom manual: 3.6 Advanced Server Features 3.6.1 Voicemail Integration SoundPoint IP is compatible with voicemail servers. The subscribe contact and callback mode can be configured per user/registration on the phone. The phone can be configured with a SIP URL to be called automatically by the phone when the user elects to retrieve messages. Voicemail access can be configured to be via a single keypress if only one registration has voicemail set up. A message-waiting signal from a voicemail server will trigger the message-waiting indicator to flash. Configuration file: ipmid.cfg For one-touch voicemail access, enable the one-touch voicemail user preference. For more information, see 4.6.1.2 User Preferences on page 65. Configuration file: phone1.cfg For one-touch voicemail access, choose to bypass instant messages to remove the step of selecting between instant messages and voicemail after pressing the Messages button (instant messages are still accessible from the Main Menu). On a per-registration basis, specify a subscribe contact for solicited NOTIFY applications, a callback mode (self callback or another contact), and the contact to call when the user accesses voicemail. For more information, see 4.6.3.5 Messaging on page 124. Now.. I've added the following in ipmid.cfg; And have the following in phone100.cfg Any idea what I am missing? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linux and Windows
On Fri, 5 Nov 2004, Michael Giagnocavo wrote: > >> >CoLinux works great for IAX to IAX or SIP to IAX where no Disk Access is > > >> >taking place. > >> > >> Thanks for clearing that up. I had been using it for IVRs. So if I > >created a > >> RAM disk for CoLinux and booted it from there... that might work? > > > >You know, that would be an interesting project, and I'd bet it would > >improve things. I'm not sure it would solve the problem entirely, but if > >you get it working, let me know. CoLinux is going through a lot of > >development work, so I expect that it's usefulness will only improve with > >time. > > How small can you get the CoLinux filesystem size? I've got 1.5GB of RAM, > but I'm imagining it'd be nice to work with less than that (esp. if you're > developing with Visual Studio on the same machine :P). About 350 megs if you keep the entire development environment and the source code around. If you go the embedded route, and build your Asterisk stuff on a different CoLinux image, and install it into the Ram Disk image, I imagine you could get the size down to about 100 megs or so. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 300 VoiceMail Retrieval
On Fri, 5 Nov 2004, Greg Boehnlein wrote: > Hello, > I have just picked up a pair of SoundPoint IP 300 phones for > testing purposes, and they work great. Really good quality units for the > price. I only have two complaints/issues with them; > > 1. The $40 additional cable for POE to work. That sucks. > > 2. I can't for the life of me figure out the "Single Keypress Voicemail > Access" that they are talking about in the manual; And true to form, a firmware upgrade fixes the problem: >From the SIP 1.3.1 release notes, a list of bugs fixed... 7570 Voice Mail cannot be accessed with standard IP 300 user interface And after the upgrade, it works like a champ. /me laments the fact that the IP 300 doesn't have a VoiceMail button. How lame! :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation
On Fri, 12 Nov 2004, Jeremy McNamara wrote: > Here is a selected portion of the strings output: > > [EMAIL PROTECTED] ~]# strings voipgw | grep Mark > Written by Mark Spencer <[EMAIL PROTECTED]> > [EMAIL PROTECTED] ~]#strings voipgw | grep CVS > Asterisk CVS-05/30/03-20:39:27 built by [EMAIL PROTECTED] on a i686 > running Linux > CVS-05/30/03-20:39:27 > Asterisk CVS-05/30/03-20:39:27, Copyright (C) 2000-2002, Digium. > Asterisk CVS-05/30/03-20:39:27, Copyright (C) 1999-2001 Linux Support > Services, Inc. [ Deleted ] > Upon confronting sysmaster with this fact, they (Mike Fahey, Ray > Martinez, and other more technical people) completely denied their usage > of Asterisk insisting they developed their solution in-house. > > I too demand sysmaster either pay Digium for a non-gpl license or > publicly admit the fact that they have repackaged Asterisk and > contribute enhancements to Asterisk back to the GPL. Wow.. this is totally unacceptable. Has anyone contacted the GNU Foundation about this? What has Sysmaster's response been? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 500 software?
On Mon, 8 Nov 2004, Rich Adamson wrote: > > That's not right. > > New phones come loaded with the current relevant firmware. > > Upgraded f/w is only available to/from certified resellers. > > Or look on the wiki for where it is freely available. > > The two new 500's that were purchased from a Polycom reseller > actually came with no firmware installed at all; only the > bootloader (or whatever its called). Someone on this list pointed > me to a souce for downloading the sip image, and now I've got the > phone running, but it won't register with *. > > Not sure what the registration problem is as yet, but doing a > sip debug indicates the registration failure. I double checked > the Auth UserID and Password and they appear to be correct. Seems > others on the list have had the same issue, but I've not found > any responses resolving the problem as yet. Anyone have any > suggestions? I just purchased a pair of SoundPoint IP 300's for use with Asterisk. Came with the SIP firmware loaded on it. Registered fine against Asterisk. I was able to register my phone at Polycom's website and setup an account to download firmware and manuals, but did not find the latest firmware there, only an older release. Found the firmware from a link on the Wiki. Loaded it, and it solved a couple of minor bugs. Phones work good. No major issues. However, I'm a little pissed that Polycom advertises the phones as supporting Power Over Ethernet, when in fact the phone has no POE chipset in it. You need to purchase an additional $40 cable if you want to plug it into a POE setup. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 300 PoE?
On Wed, 17 Nov 2004, Noah Miller wrote: > I'm ordering some more phones - I have the Polycom IP 500's now and I > like them. I need some less expensive phones, and I'd like to stay > with all Polycoms for ease of administration. I've heard, though, that > the IP 300's don't support PoE even though their brochures say they do. > Has anybody have firsthand experience with them? Is this true? Yep. They have no onboard POE chip, hence the reduced cost. Polycom gets away saying they support POE by selling you at $40 cable that contains either a Cisco or 802.3af compliant dongle. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] PoE switch question (Netgear FSM7326P works with Cisco)
On Thu, 18 Nov 2004, Kevin P. Fleming wrote: > So, I ask again: given the choice between a sub-$100 16-port full-duplex > 100Mb switch and external power supplies, and an over-$1000 12-port > switch with internal power supply, which do you think is a better value > for a small LAN? I can buy $20 3Com PoE bricks and hook them all up to a > UPS for a lot less than $900, with the downside being that it will be > ugly to look at (and the bricks aren't "real" PoE, but they are close > enough for VOIP phones). Well, that depends how important it is to have phone service during a power outage, and what their UPS budget is! ;) When you add another $55 / workstation for individual UPS units just to ensure the phones work, rather than centralizing that and getting a big-ass central UPS for the entire system, the numbers even out a bit more! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Just getting started...
On Sat, 20 Nov 2004, Wilson Pickett wrote: > > Here's my current plan: > > > > Sounds like a plan? > > You asked for advice, here comes some that few will approve of :) > > FWIW I tried to get gnophone running and got no further than you did. > What struck me though was that I have a very linux wise programmer > friend and associate that never got it running either. > > The unpalatable advice of mine is that for initial testing, it would > be good if you could either have a Windows box or laptop to avail > yourself of the larger number of softphones, among them a few that > work pretty well, or, bite the bullet and buy an IAXy if you can > afford it and feel you'll be investing in one later. Or try SIP just > to get things running. If you decide to go down the Windows client route, I would highly reccomend Virbiage's FireFly IAX client. I use it quite a bit on my Laptop and it has been rock solid. BTW.. I'm glad you liked the Presentation! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot get two TE410Ps to operate correctly in the same machine
On Fri, 26 Nov 2004, Rich Adamson wrote: [SNIP] > What if you have a single port T1/E1 card from digium? No decision to > be made really; you're going to sync from the other end if it goes > higher in the hierarchical chain. If the port goes to a box consider > lower in the chain, then the distant box should be configured to > accept clock sync. How about the following topology for a bit of discussion: Telco -> T100P -> * Server A -> IAX2 -> * Server B -> T100P -> PRI PBX In this case, we have Server A pulling timing from the telco, ensuring they are in Synch. On the other side of the IAX2 link, we have Server B generating timing onto the span for the PBX to listen to. T100P is set to emulate an NI-2 switch. Seems to work without much of a problem, as it should! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Business Weekly Article
My copy of VoIP Business Weekly came today and the main article on the front page is titled: "Penguin to Assault PBX Market". Their premise is that Linux is driving the adoptation of Open voice standards in the market, and as such is poised to disrupt the dominance of market leaders. Third paragraph is a description of Asterisk and and the laundry list of features. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pre-installation jitters
On Wed, 1 Dec 2004, Samudra E. Haque wrote: > I would like to build my newest server based upon Fedora Core 3, and > load up asterisk. I was all set to do so.. but then I read in Asterisk > Users Digest, Vol 4, Issue 404: I think you would be insane to run your production servers on Fedora Core 3. Most of the components in Fedora are experimental and not well tested. Indeed, the entire point of Fedora is to experiment. You would be much better served to run your installation on RedHat Enterprise Linux, or one of the free alternatives such as Tao Linux or White Box. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sveasoft Alchemy QOS
On Wed, 1 Dec 2004, Kanuri, Seshu (Company IT) wrote: > > > Tell me which one can get me access to the LinkSys Linux using SSH? > > Does Satori has this feature? I am not so concerned with Voice Shaping > > > and QOS at this time, but more interested in converting this into a > > Linux box that is accessible from an ssh client. > > Alchemy has ssh access, you need to pay $20 subscription to Sveasoft to > access the pre-release firmware. > > Steve > > - > > $20.00 for GNUed "hackware" that is originally freely donated by > LinkSys? No way. Well, then roll your own and stop whining about it. Quite frankly calling it "hackware" shows that you have no concept of how much work has gone into the Sveasoft firmware, nor do you grasp the concept that Linksys is incorporating many of the Sveasoft changes BACK into their firmware. Everyone wins from this, and Sveasoft has a revenue stream that allows them to keep focused development on improving the firmware. I have over 60 of the WRT54GS units in production and I run Sveasoft firmware on every single one of them. It is so far ahead of Linksys's internal builds and adds so many additional features that there is no comparison between the two. "Hackware" indeed. What an insult to all of the quality developers that are putting their time and effort into extending the platform and making it one of the most incredible sub $100 routing platforms on the planet. > How much did I pay for Asterisk? Was it $20 grand? > Don't remember having paid that much for Asterisk. $20 US DOLLARS! Not, $20,000! > If not SSH what other way can we access Linksys Linux without Alchemy > wrapper sftp/rlogin/telnet? Does anyone know? Plenty of people know. If you can't do a google search for it, this isn't the place to be asking about it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ring all Configured Extension
On Thu, 2 Dec 2004, Eric Rees wrote: > Where only talking about 100 extensions. That is a lot to hard code by > hand. Just use app_queue and define a list of members as the SIP extensions. It is a lot easier to maintain the queues.conf file than to worry about adding 100 extensions into your dial-plan. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matthew > Boehm > Sent: Thursday, December 02, 2004 12:12 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Ring all Configured Extension > > Why are you afraid of that suggestion? > > Matthew > - Original Message - > From: "Eric Rees" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, December 02, 2004 10:56 AM > Subject: RE: [Asterisk-Users] Ring all Configured Extension > > > I was afraid that someone would suggest that. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matthew > Boehm > Sent: Thursday, December 02, 2004 10:06 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Ring all Configured Extension > > exten => 4000,1,Dial(SIP/3001&SIP/3002&SIP/3003&..., 30, > t) > > Matthew > - Original Message - > From: "Eric Rees" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, December 02, 2004 8:56 AM > Subject: [Asterisk-Users] Ring all Configured Extension > > > I don't know if the is possible on not. I would like to know the > easiest way to ring all extensions in the sip.conf file for intercoms. > I have phone to phone intercom working. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
On Sat, 21 Aug 2004, Ed Devine wrote: > I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that > I can experiment with. I've been wanting to use Redhat with software > Raid 1 on an Asterisk server. > > Has anyone had any experience with software raid and Asterisk? Also, if > the software raid doesn't play, any recommendations for a hardware based > IDE Raid controller and suggestions on best practices for setting up the > disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be > appreciated. I have a Dell, 1 Ghz PIII 1U rackmount server, w/ a T100P (PRI) running Tao Linux (RedHat Enterprise 3.0) on 2 40 gig drives in RAID-1. I've had as many as 20 calls on the box at one time, and never noticed an issue. Of course, 90% of our calls are pass-through Ulaw, and although we have g729 licenses it doesn't get used very much. Never had a problem with it so far. However, if you are looking for an IDE Hardware Raid solution, I have tried everything on the market, and I can honestly say that for price, compatibility and reliability, the 3Ware cards are the best bet. This is the only card that we spec for clients, as the Promise and Adaptec stuff is garbage. Incidentally, if you want a good deal on a Promise Fast-Track SX 6000, I have a few laying aroung! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
On Sat, 21 Aug 2004, Lyle Giese wrote: > I used software raids and want to get away from them. I really really > like Microlite's BackupEdge tape backup software. BackupEdge does not > work with a software raid, only a hardware raid. > > The second kicker was that the Promise (or any other) hardware IDE raids > are considered software raid to the Kernel. And are NOT supported with > the new 2.6.x Kernels... This is NOT generally true. Many of the embedded Promise controllers, built into the motherboards, are detected as separate devices. > My experience with IDE in raid arrays is less than stellar and will be > trashing them as I rebuild servers. I have had several instances where > one drive fails and the entire array falls over as the kernel struggles > to recover from the loss of a drive or the error messages. I have seen > this with Linux generated arrays and with the Promise IDE raid cards. > Besides the performance of the Promise parallel IDE raid sucks big time. Well, the Promise cards suck in general. That's not Linux's fault! :) > With the excellent tape software above, I don't think I really need > them anyway. BackupEdge will generate a bootable CD-Rom that will > install your last backup on bare metal after a major malfunction. Here is another vote for MicroLit's Backup Edge. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and software Raid
On Mon, 23 Aug 2004, William Boehlke wrote: > Consider hot swappable SCSI RAID 1 instead of IDE. You'll appreciate it > every couple of years when you lose a disk but the PBX stays up. I hot swap on my 3Ware IDE RAID card all the time. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed
On Fri, 27 Aug 2004, Andrew Brown wrote: > > When using the callback feature on agents I notice that when the queue calls > one of the agents and the agent picks up the call they hear nothing until > pressing the # to accept the call. > > Only then does my announcement play back to the agent after which the call > is immediately connected. > > Is there a way to have the announcement played to the agent before they > press # to accept. I have ackcall=yes in agent.conf > > Can't find anything on the wiki. > > Thanks > > Andrew Andrew, You need my agent-preack-announce patch. This allows you to define a "preack=gsmfile" in your agents.conf, which will be played BEFORE the agent hits the "#" key to accept the call. The patch and associated sound files etc.. are up on the bugtracker at: http://bugs.digium.com/bug_view_page.php?bug_id=0001082 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POE
On Sat, 28 Aug 2004, Michael Welter wrote: > Since the Cisco 79XX phones preceded the PoE standard, they are > different--polarity is reversed. > > IANAE, but as I understand the PoE devices, there are two types--one > always applies -48VDC to the brown pair while the other senses (as per > the PoE spec.) whether the device at the other end requires power. > > I'm not willing to risk a $300 Cisco set, so I'm still using the wall > wart. Is anyone providing LAN power to 79XX phones at a reasonable cost? Please see: http://www.voip-info.org/wiki-Cisco+POE I am using 3Com POE injectors (the Dumb kind) with Cisco 7960s without a problem. You need a special cable that you can build or, you can buy a PowerDSine POE injector and get the following (part PD-PS-401-5/CSCS) which will do the same thing as the cable. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POE
On Sat, 28 Aug 2004, Michael Welter wrote: > Greg Boehnlein wrote: > > On Sat, 28 Aug 2004, Michael Welter wrote: > > > > > >>Since the Cisco 79XX phones preceded the PoE standard, they are > >>different--polarity is reversed. > >> > >>IANAE, but as I understand the PoE devices, there are two types--one > >>always applies -48VDC to the brown pair while the other senses (as per > >>the PoE spec.) whether the device at the other end requires power. > >> > >>I'm not willing to risk a $300 Cisco set, so I'm still using the wall > >>wart. Is anyone providing LAN power to 79XX phones at a reasonable cost? > > > > > > Please see: > > > > http://www.voip-info.org/wiki-Cisco+POE > > > > I am using 3Com POE injectors (the Dumb kind) with Cisco 7960s without a > > problem. You need a special cable that you can build or, you can buy a > > PowerDSine POE injector and get the following (part PD-PS-401-5/CSCS) > > which will do the same thing as the cable. > > > > To add to my confusion, the wiki page you referenced says the 7960g > phones are different from the original 7960. Not in terms of POE. No 7960 supports the 802.3af POE protocol and instead uses a proprietary Cisco protocol. The cable that is referenced on those pages reverses the polarity of the wires, which with a dumb power injector like the 3com will allow you to use the Cisco. However, if you get a regular 802.3af injector, that actually speaks and respects the protocol, it STILL won't work with a Cisco. I use the following dumb injector with my Cisco cable; http://www.3com.com/prod/en_UK_EMEA/detail.jsp?tab=features&sku=3CNJPSE It works. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Migrating Asterisk
On Wed, 1 Sep 2004, Jay Milk wrote: > Hello All, > > My asterisk installation has now been running for over two months > without a hitch, and I've decided it's time to move things around a bit. > It's currently installed on a 2.7GHz Celeron under RH9 installed on a > 10GB "leftover" drive. Thanks to the strange marketing method called > "Mail-In-Rebate", I have a fresh 160GB drive ($50), and I'm itching to > install a GenToo Linux distro. > > I also have a 1.2ish GHz Duron with Mobo sitting around here, which may > just be enough to power my (barely ever transcoding) asterisk install. > Should be enough, even if one channel were transcoded occasionally, no? I run Asterisk on a Pentium 133 w/ 16 megs of ram and routinely have 2-3 ulaw to g726 transcoded sessions going on. Works just peachy. The max I've ever done is 6 channels before I started having some dropouts in audio. So a Duron 1.2 Ghz should do just fine. ;) > Let's say I start with a fresh machine, GenToo (2.4 Kernel), and a > recent Asterisk (which one? I'm running HEAD from 05/02/2004 right now, > heavy on SIP, no problems), and move one of my two X100P for the timing > source... Would it be enough to copy over the Asterisk config and VM > files? (yes, yes, they'll share IP addresses, so I don't have to > reconfigure my devices) > > So... > 1) 1.2Ghz Duron, enough for transacoding a single channel? > 2) X100P sufficient timing device for *? > 3) Which * source does the list recommend? > 4) \var\lib\asterisk, \etc\asterisk and zaptel.conf are all that's > needed to migrate the current state of *? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
On Wed, 8 Sep 2004, hank smith wrote: > hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for > windows and am having trouble getting the thing to connect to the meers > to download the updates and stuff. I looked at the wiki and set up > networking and stuff with no success, has any one got this thing to work > successfully? Yes. The networking issue that you are fighting with is the hardest part. If you haven't please check out: http://www.colinux.org/wiki/index.php/coLinuxNetworking Look about halfway down the page for "Native network (Bridged) (Windows2000 or WindowsXP)" Follow the instructions. I'll bet any money that you are having problems finding out what the real name of your Network adapter is. > my windows box is the faster of the 2 machines and my main linux box is > down at the moment. I am running a netgear rp614 router behind nat if > this helps but I have tried and tried and tried to get this sucker up > with no luck any help would be greatly greatly appreciated. > thanks > hank Come grab me on irc.freenode.org #asterisk.. Look for Damin. I put together the installer, so I can certainly help! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
On 9 Sep 2004, khurram bhatti wrote: > Well I wanted to test astwind and consulted * person > he gave me this comment > "lord help us all ... why would you want to simulate a linux system on > top of a windows system in the first place?" It's not a "simulated linux system". CoLinux is a kernel that runs in Ring 0 of the Windows kernel, with direct access to the Processor and MMU. It runs in it's own protected memory space. The ONLY thing it uses Windows for is to actually load the kernel and handle the I/O drivers. Otherwise, CoLinux is running natively on your hardware... At the same TIME as windows. So, it isn't like running Vmware. It's a LOT faster, and if you set it up right, you can even boot your existing Linux partition. It's the best way to run Windows -AND- Linux at the same time. Blows the pants off of BoCHS and Vmware in speed. I've been running my home PBX under AstWind for about a month now. Even after the Windows kernel has crashed and the system is completely hung, CoLinux and AstWind continue to run without a problem! It's pretty amazing. Check out http://www.nacs.net/~damin/astwind.jpg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astwind has any one got this thing to work?
On Wed, 8 Sep 2004, Chris HARIGA wrote: > I make it work!! > > My Astwind is up and running! > Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix > the Ethernet connection. I bet you followed the following directions! ;) From: http://www.colinux.org/wiki/index.php/coLinuxNetworking "If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is "NDIS 5.0 driver" and not "Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter". It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta" -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astwind has any one got this thing to work?
On Thu, 9 Sep 2004, Chris HARIGA wrote: [SNIP] > > > Yep.. Don't you hate the network driver names that these things pick? Couldn't they just call it "Intel PRO 100" or something simple like "Pro 100" CoLinux 0.6.2 will allow you to use the name that YOU assing to the connection in Windows. In most cases, this defaults to "Local Area Network", but you can easily change this by right clicking on it. The next version of AstWind will use CoLinux 0.6.2 and will be far easier to setup. I may even let you choose a network adapter in a pull down from the installer and write that to the XML file for you upon installation. If you have any tips, suggestions or please feel free to Email me, or contribute your information to the AstWind Wiki page: http://www.voip-info.org/wiki-AstWind -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
On Thu, 9 Sep 2004, arsal siddiqui wrote: > dear khurram, > > i need to know the price for x100p. i've emailed convergence.com.pk > and never get a reply. If you could help me in this regards, i'll be > greatful. I need to know the price. > > send me an email off the list. if you can help me in getting * hardware. > > Waiting for your reply Just as a side note... CoLinux CANNOT YET interface with any Digium hardware! So if you plan to run an X100P under AstWind you may be waiting a long time before it works! ;) Someone needs to port Zaptel to CoLinux! ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to post SuSE 9.x startup script?
On Thu, 9 Sep 2004, Martin Mielke wrote: > Hi all, > > due to the rather big email traffic regarding this issue, I decided to > publish the script so people can download it at their own risk... :-) > > Please, visit: > > http://www.leals.com/~mm/asterisk > > for further information. > > Regards, > Martin Martin, I authored both the rc.debian.asterisk and rc.redhat.asterisk scripts that are currently in CVS. Your script looks good and works. However, would you mind taking a look at the rc.redhat.asterisk script in your asterisk/contrib/init.d directory and comparing it against yours? Both the Debian and RedHat scripts in CVS have some intelligence built into them that do the following: 1. Check for, and use safe_asterisk if it is available. 2. Allow for the setting of the User and Group that Asterisk will run under using the new "-U" and "-G" options that Mark added. 3. Correctly detect and pass these parameters to safe_asterisk, which as of June, was modified to append command line arguements. If you do not have the time to do this, I'd like permission to take your work, merge it with mine and submit it to CVS. I've volunteered to handle the init scripts for the various platforms. Let me know.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 can't conference.
On Thu, 9 Sep 2004, Matt - Telcom Products wrote: > Hello, > Does anyone know how to conference a call on the SNOM 200 phone? >Whenever I push the cnf/tran button it just hangs up on the active call. >The manual says you have to push the cnf function key but it doesn't >appear in the lcd on my phone. > Thanks > -Matt First, update to the following firmware: http://www.snom.com/download/share/snom200-3.49-SIP.bin Second, the cnf hard button doesn't work. Use the SOFT key that shows up on the LCD that says "Cnf On". I.E. put the first party on hold, call the second party, and after they answer the "Cnf On" button will appear in the LCD. Or, you can just Xfer both parties to a MeetMe and call into it yourself. I do this a lot. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
On Thu, 9 Sep 2004, hank smith wrote: > I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that > what I put in the xml file? Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking Specifically: "If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is "NDIS 5.0 driver" and not "Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter". It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta" Deja Vu.. Is there an echo in here? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
On Thu, 9 Sep 2004, hank smith wrote: > is there going to be a gui for co linux and astwind? No. AstWind is just a Debian GNU Linux distribution with a precompiled Asterisk installation running under a CoLinux kernel. > I will have to see if either there is going to be a gui or if yasr a screen > reader for the blind will work with this thing. I do not know. I would assume that a blind user would probably prefer a text based interface, but I have no clue. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
On Fri, 10 Sep 2004, Benjamin on Asterisk Mailing Lists wrote: > However, you could use VMware on an Intel notebook to run both Windoze > and Linux concurrently. This wouldn't be ideal for a real PBX for > performance reasons, but since all you are going to use Asterisk for > is to be a gateway for one single user, it's probably ok in this > particular scenario. Or you could use AstWind, which runs concurrently with Windows and is built entirely on Open Source software (CoLinux Kernel, Debian, Asterisk) and avoid paying for Vmware! ;) Plus, installation is a snap. See Digium's press release: http://www.digium.com/index.php?menu=astwind You can find more information on AstWind at: http://www.voip-info.org/wiki-AstWind -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk presence utility
On Sat, 18 Sep 2004, Bill Seddon wrote: > I've spent a couple of evenings writing a "presence" utility in C# so that a > window, listing the currently registered SIP phones, can be displayed and a > user can see who is on the phone and who is not. It uses the Manager API > and works well, updating the display as API events are received. > > But what I want to be able to do is add to the list of displayed users when > a phone registers and remove them when a phone unregisters. However the > Manager API does not seem to generate event messages for these events. Is > this correct or have a I missed an option somewhere? Certainly the register > and unregister event is displayed on the Asterisk command line. > > It is possible to have the utility run the "sip show peers" command > periodically and update its list based on the results of the command. > However that means polling the server periodically, comsuming resources, > instead of being notified just once. > > Any insight gratefully received. > > Bill Seddon This is purely conjecture on my part, but it seems to me if the console is generating SIP Un/Registration messages, that it would be trivial for the manager interface to display them as well. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0 Mirrors
Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 -- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 -- ftp://ftp.nacs.net/asterisk/rh9/RPMS/ ftp://ftp.nacs.net/asterisk/rh9/SRPMS/ Changelog - * Thu Sep 23 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated to version 1.0 - Drank beer at Astricon * Thu Aug 12 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated to version 1.0-RC2 - Replaced parking.conf with features.conf * Sat Jul 17 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated to version 1.0-RC1 * Thu May 27 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated to version 0.9.0 * Wed Feb 04 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated to version 0.7.2 * Sat Jan 31 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated development environment to ensure proper build consistency for chan_zap - Added post-install chkconfig to auto-start asterisk on boot - First really useable release. Yay! * Mon Jan 26 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated changelog entry to enable build on Fedora Core 1 <[EMAIL PROTECTED]> - Made the decsision to use Dist Specific version numbers (_fc1,_rh9,_rh8,_rh73) * Sat Jan 24 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - added doc macros - added config macros - updated install stanza to correct symlink issue - updated patch0 to include changes to Makefile - added /etc/rc.d/init.d/asterisk - added "export LD_ASSUME_KERNEL=2.4.1" for RH9 - asterisk.spec now builds cleanly on RH73 and RH9 * Wed Jan 21 2004 Gregory J. Boehnlein <[EMAIL PROTECTED]> - Initial .spec file created. Most likely buggered. Badly needs help. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Thu, 23 Sep 2004, Gary Carr wrote: > The RPMs had errors for me > > > After installing RPMS and running modprobe zaptel I get > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > register_chrdev_R07a6f6f0 > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > remove_wait_queue_Rd7b46182 > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > remove_proc_entry_R16f1fe81 > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > __pollwait_Rb9575694 > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > proc_mkdir_R68919af9 > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > create_proc_entry_Rd11cc972 > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > add_wait_queue_R1891d4b7 > /lib/modules/2.4.20-31.9/misc/zaptel.o: insmod > /lib/modules/2.4.20-31.9/misc/zaptel.o failed > /lib/modules/2.4.20-31.9/misc/zaptel.o: insmod zaptel failed > > > Going back to downloading directly. Gary, When you do a "uname -a" what kernel version are you running? I'll take a look at this after we get back from the Waffle House. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Fri, 24 Sep 2004, Greg Boehnlein wrote: > On Thu, 23 Sep 2004, Gary Carr wrote: > > > The RPMs had errors for me > > > > > > After installing RPMS and running modprobe zaptel I get > > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > register_chrdev_R07a6f6f0 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > remove_wait_queue_Rd7b46182 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > remove_proc_entry_R16f1fe81 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > __pollwait_Rb9575694 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > proc_mkdir_R68919af9 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > create_proc_entry_Rd11cc972 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > add_wait_queue_R1891d4b7 > > /lib/modules/2.4.20-31.9/misc/zaptel.o: insmod > > /lib/modules/2.4.20-31.9/misc/zaptel.o failed > > /lib/modules/2.4.20-31.9/misc/zaptel.o: insmod zaptel failed > > > > > > Going back to downloading directly. > > Gary, > > When you do a "uname -a" what kernel version are you running? I'll > take a look at this after we get back from the Waffle House. Anyone else having the problems that Gary is reporting? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Fri, 24 Sep 2004, John Bohman wrote: > Same problem here.. > Redhat 9 2.4.20-31.9 John, Let me look into it, and rebuild the RPMS. Perhaps I have a header missing issue. > John B > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein > Sent: Friday, September 24, 2004 8:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9 > > On Fri, 24 Sep 2004, Greg Boehnlein wrote: > > > On Thu, 23 Sep 2004, Gary Carr wrote: > > > > > The RPMs had errors for me > > > > > > > > > After installing RPMS and running modprobe zaptel I get > > > > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > register_chrdev_R07a6f6f0 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > remove_wait_queue_Rd7b46182 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > remove_proc_entry_R16f1fe81 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > __pollwait_Rb9575694 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > proc_mkdir_R68919af9 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > create_proc_entry_Rd11cc972 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol > > > add_wait_queue_R1891d4b7 > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: insmod > > > /lib/modules/2.4.20-31.9/misc/zaptel.o failed > > > /lib/modules/2.4.20-31.9/misc/zaptel.o: insmod zaptel failed > > > > > > > > > Going back to downloading directly. > > > > Gary, > > > > When you do a "uname -a" what kernel version are you running? I'll > > take a look at this after we get back from the Waffle House. > > Anyone else having the problems that Gary is reporting? > > -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Sat, 25 Sep 2004, Florin Andrei wrote: > On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote: > > > Anyone else having the problems that Gary is reporting? > > Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel > 2.6) and i had to add a "linux 26" at the end of the "make" line, > otherwise all kinds of weird things happened. I think I'll add a kernel version check to it and either force a "make" or a "make linux26" based on what is running. > Also, in /etc/init.d/zaptel, "insmod" doesn't work properly. It has to > be replaced with "modprobe". I have no idea why. > > There are some other changes i've made to the initialization scripts, to > bring them closer to Red Hat "best practices". I'll probably email you > privately when i'm closer to a stable state. On the init.d front, Mark committed my much improved redhat init scripts to CVS last month, but I haven't updated the RPMS to use them. > Anyway, the RPMs are way cool! :-) Thanks, You are welcome. Keep in mind that any changes that you make should take into account the original design goal, which is to have a universal .spec file for all versions of RedHat from 7.3 onward. I.E. please try to keep distro specific changes defined ONLY for that distro and not globally applied to everything. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Simple Manager Proxy
On Sat, 25 Sep 2004, David Troy wrote: [deleted[ > I had a need for a much simpler proxy than his op_server.pl; to meet my > need I re-worked and simplified his code. See below for this simplified > proxy: > > http://www.popvox.com/simpleproxy.pl Hehehehe.. I mentioned this in the Developer's Session at Astricon.. that op_server.pl might form the basis for a middleware to interface many clients w/ Asterisk for all sorts of things.. presence, management etc.. See how great minds think alike? ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Mon, 11 Oct 2004, Pete Brown wrote: > Greg, > Which kernel are you using? I have two machines at home and the zaptel > kernel module only runs properly on one of them... > > The P-3 box worked... > kernel-2.4.20-30.9.i686.rpm RH73 Zaptel Modules are built against: 2.4.20_28.7.i386 RH9 Zaptel Modules are built against: 2.4.20_31.9.i386 > The Athlon did not... > kernel-2.4.20-31.9.athlon.rpm > > Both machines were updated on the same day (apt-get) and for the most part > have the same packages. Has anyone made any headway on this? This is one of the issues w/ Zaptel. It has to be built for a specific kernel and architecture, so without maintaining every possible combination of kernel and OS, you are actually better off to just build the SRPM on your reference development platform. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 availability
On Mon, 11 Oct 2004, Mark Phillips wrote: > I had a rather unpleasant bait and switch episode with Atacomm today. > They advertise on their website (and indeed quoted me for) the Snom 200 > for $269 which, when I came to place an order for 15 of them, they > didn't have but would like to replace with the 220 at $379. > > They came up with some crap about Snom not shipping the 200 currently to > the US but that I could have them in January. Has anyone heard this to > be the case? What about other suppliers? I talked with the guys at Atacomm, and NetxUSA about the SNOM 200. Apparently, SNOM has temporarily stopped production of the SNOM 200 model, and instead replaced it with the SNOM 190. The problem is that the SNOM 200 supports POE, but the 190 does not. SNOM claims that they will be manufacturing more SNOM 200 units. I certainly hope so, as the SNOM 200 is a great phone. All other things aside, if I'm going to spend $269 on a phone though, it had better support Power Over Ethernet. So, SNOM, if you are listening, please keep in mind that the PolyCom SoundPoint IP-500 is right in your market space, has a great speakerphone, supports POE and is cheaper. If you are going to produce more 200's, consider dropping 30% off the price, or you just might find yourselve's shut out of the market. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 200 availability
On Mon, 11 Oct 2004 [EMAIL PROTECTED] wrote: > Everyone: > > We are a Snom authorized reseller and the problem with the Snom 200 is the > fact that Snom has EOL that model. It is being replaced with the Snom 190. > The reason there are no Snom 200's is these unit were taken out of > production approximately three months ago. That leave's many reseller's in > short supply (or none). The Snom 190 was just released this month, and is > the replacement for the 200 in Snom's phone line up. > > Here is a link to the Snom 190 > http://www.voipsupply.com/product_info.php?cPath=3_55&products_id=260 Except that it doesn't support Power over Ethernet, so it isn't a replacement at all. > Regards, > > Garrett Smith > B2 Technologies > <[EMAIL PROTECTED]> > www.voipsupply.com -Your One Stop VoIP Shop- > www.valueresale.com -For All of Your IT Needs- > > > > > Try ABP Tech www.abptech.com > > > > Regards > > > > HA > > > > -Mensaje original- > > De: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] En nombre de Mark > > Phillips > > Enviado el: Monday, October 11, 2004 6:51 PM > > Para: [EMAIL PROTECTED] > > Asunto: [Asterisk-Users] SNOM 200 availability > > > > I had a rather unpleasant bait and switch episode with Atacomm today. > > They advertise on their website (and indeed quoted me for) the Snom 200 > > for $269 which, when I came to place an order for 15 of them, they > > didn't have but would like to replace with the 220 at $379. > > > > They came up with some crap about Snom not shipping the 200 currently to > > the US but that I could have them in January. Has anyone heard this to > > be the case? What about other suppliers? > > > > > > -- > > > > Mark Phillips, G7LTT/KC2ENI > > Randolph, NJ > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS Router/Software Suggestions
On Tue, 12 Oct 2004, Matthew Boehm wrote: > Switching to DSL would require me to get a phone line, which kinda defeats > the purpose of doing VoIP. =) > > Matthew Matthew, for unparalelled hackability try the Linksys WRT54GS. It runs Linux, and supports QOS if you use the Sveasoft (http://www.sveasoft.com) firmware, or the OpenWRT firmware (http://www.openwrt.org). I use the Sveasoft firmware ($20 / year for a subscription), but it kicks complete ass. I don't need to worry about anything, or you can build it yourself. Also, people have Asterisk actually running -ON THE ROUTER- itself. How's that for cool? ;) http://www.linksys.com/products/product.asp?grid=33&prid=610 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QoS Router/Software Suggestions
On Tue, 12 Oct 2004, Geoff Nordli wrote: > Is this where we get to vote for our favorite router software? I choose > Bering-uClibc > (http://leaf.sourceforge.net/mod.php?mod=userpage&menu=910&page_id=36). It > comes with a ton of packages, and you can easily configure it to boot from > HDD, or Compact Flash. Of course it also comes with QOS/Traffic Shaping > support. Plus all the VPN options (IPSEC, PPTP, OpenVPN). > > I have been thinking before about adding an * package to it so you could > deploy it remotely and not worry about SIP problems. I have heard there are > problems building Asterisk with uClibc. Well, yeah. ;) But.. on another note, I just had what could amount to a brain-fart, or a good idea depending on how you look at it. There are some big issues with getting Asterisk to compile using uClibc rather than Libc, but if we could take the initial step of actually getting CVS Head to build cleanly against uClibc, these patches could be integrated back into the source tree making Asterisk more portable to embedded platforms. Sure, running uClibc on a Soekris or Via which have MMU's is not the same as a MIPS w/out an MMU, but it would be an important first step. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Svar: Re: [Asterisk-Users] Where to post SuSE 9.x startup script?
On Mon, 18 Oct 2004, Claus Lavdal wrote: > I would be very interested in the script that allow me to use > saft_asterisk and non-root user on suse 9.1. > > Regards Claus Claus, Sorry it took a couple of days to get this information back to you, but I wanted to let you know that you can find the rc.redhat.asterisk and rc.debian.asterisk scripts in the asterisk/contrib/init.d/ directory. You can modify either one to suite for SuSe.. > >>> [EMAIL PROTECTED] 10-09-2004 06:22:23 >>> > On Thu, 9 Sep 2004, Martin Mielke wrote: > > > Hi all, > > > > due to the rather big email traffic regarding this issue, I decided > to > > publish the script so people can download it at their own risk... > :-) > > > > Please, visit: > > > > http://www.leals.com/~mm/asterisk > > > > for further information. > > > > Regards, > > Martin > > Martin, > I authored both the rc.debian.asterisk and rc.redhat.asterisk > scripts that are currently in CVS. Your script looks good and works. > However, would you mind taking a look at the rc.redhat.asterisk script > in > your asterisk/contrib/init.d directory and comparing it against yours? > > Both the Debian and RedHat scripts in CVS have some intelligence built > > into them that do the following: > > 1. Check for, and use safe_asterisk if it is available. > 2. Allow for the setting of the User and Group that Asterisk will run >under using the new "-U" and "-G" options that Mark added. > 3. Correctly detect and pass these parameters to safe_asterisk, which >as of June, was modified to append command line arguements. > > If you do not have the time to do this, I'd like permission to take > your > work, merge it with mine and submit it to CVS. I've volunteered to > handle > the init scripts for the various platforms. > > Let me know.. > > -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DUNDi in stable? (New subject)
On Tue, 19 Oct 2004, Olle E. Johansson wrote: > > I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty > > simple, or do the code changes run deep? > It will eventuall get into a release. > > But please, as a community, we have to refrain from temptation of adding > new functions to the stable tree. It has to be kept stable - and that is > boring. If you want to walk on the wild side, run CVS head. > > CVS head is the development tree. Please don't encourage people to use it > in production environments, even if it from time to time seems to work well. > We need to be able to include new untested functions. Some days, it doesn't > compile properly after you download it to your system. That is okey, it means > that we have new code, new functions and new bugs to fix. > > We need a development CVS as fertile soil for new major Asterisk breakthroughs. > I've been waiting long for that to happen, and if you check the -cvs list, > you'll find that there's been a lot of changes to the development branch > that would never have happened if we didn't have a less strict environment > to play around with. Keeps the bug tracker alive :-) > > /Olle Amen! Speak the truth brother Olle! :) I, too, have been waiting for a Stable tree. Sure, it's boring, but enterprising people can always backport features and add them as stand-alone modules to asterisk-addons. I am using Dundi with my 1.0.1 servers and it works. Ask on #asterisk for the modularized version of it and someone is bound to point it out. Simply "make install" and grab the "dundi.conf.sample" from current CVS and go to town. However, now that 1.0 has been feature frozen and released, there are a whole lot of companies out there who have breathed a large sigh of releif and are polishing up the final releases of ther Asterisk related software. This helps EVERYONE in the community, and further drives Asterisk into new locations. It allows Asterisk to actually have releases which are a known quanity and can be picked up and added to the various distributions. This is a GOOD GOOD thing! :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 5,000 concurrent calls system rollout question
On Thu, 2 Feb 2006, John Todd wrote: [SNIP] > 3) Nobody else has thus far taken the bait and made any comments > about their systems. I appreciate Signate's comments; they seem to be > the only ones to publicly claim large-scale throughput using Asterisk > in a public forum. Most other people who claim thousands or even > high hundreds of connections do so offhand, without responding to > second questions when I raise my figurative eyebrows. John, Per our conversation in San Fransciso, I am starting to push a couple of my Asterisk boxes farther than I've gone before. I'm not yet anywhere near the 5,000 concurrent call level on my boxes, but I am starting to see 150-160 concurrent calls coming through the system. In this case, these are SIP to SIP where Asterisk is staying in the media stream, but rarely transcoding. Approximately 99% of the calls coming through are just pass-through g729, with the occasional gsm conversion. I'm running Asterisk 1.2.4-svn in a completely stock configuration. I.E. no patches whatsoever, and absolutely performance tweaks. In fact, the system is running using MALLOC_DEBUG to catch memory leaks and is built using "dont-optimize" so we can get backtraces if things go south. My Dial-Plan is highly optimized, with a focus on being as efficient as possible while offering failover options for call completion. > 4) There are still no notes on other problems with scale here. I've > had systems with several hundred simultaneous SIP connections, but > "sip show channels" sure does start to take a while. What _other_ > problems crop up, but don't necessarily cause a "failure" condition? Well, debugging anything on the console with 160 concurrent calls coming through the system (sometimes 4-5 calls / second) is nearly impossible. Most of the time, I don't even run the console, and simply execute commands from a bash prompt as "asterisk -rx 'sip show channels'". I ALWAYS, ALWAYS, ALWAYS issue a "set verbose 0" before I reload the box, as a reload causes the box to hiccup slightly while it is printing the data to the console. I had originally opted to write CDRs to disk and then import them into a SQL database, but after I cleaned up my dial-plan, I opted to use cdr_odbc. I am concerned that this could cause a blocking condition if the SQL server is unavailable, but for now I'm taking the risk because I need to have real-time stats on call statistics. > 5) I will agree that most SIP testing systems are currently too > pricey. I would love to find a well-connected network that rents out > a few of the better-known SIP testing tools to beat on Asterisk > installations in remote places for short periods of time. But this > has always been the case... test gear is a small market, and > expensive. Just look at the MSRP of new high-end HP Oscilloscopes if > you want to get a picture of price-gouging. I know that Olle spent some time at SipIt w/ Asterisk, and he's been interested in doing some additional compliance and scalability testing. I'd like nothing better than to get a couple of key developers together for a weekend of scalability bashing somewhere, preferably outside of the regular conference circuit (too distracting) to push things to their limits. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth?
On Wed, 29 Jun 2005, Paul Fielding wrote: > I have indeed already done so - I use G729 quite a bit since I travel alot > in adverse conditions. Interesting thing is, I can get less choppy audio > sometimes from my Vonage device using (what I suspect to be) Ulaw, while > either ulaw or G729 will still give choppy response at that moment from my > Linksys > > Paul Funny thing about codecs.. Sometimes you will get better quality audio and less chop when using ulaw on crappy connections. Why? Very simply because it sends less total audio information per packet, so it can withstand dropped packets a lot better than a highly compressed codec, especially with a good jitter buffer and PLC implementation on the other side. I'd expect that Vonage is running something with a decent PLC and Jitter buffer, which would explain your results. Try using CVS head and enabling the SIP jitter buffer and PLC code w/ Ulaw and see if it improves your results. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Memory Leak in Stable?
Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been no-one in the office placing any calls. [EMAIL PROTECTED] tftpboot]# asterisk -rvc Asterisk CVS-v1-0-07/03/05-05:09:22, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <[EMAIL PROTECTED]> = Connected to Asterisk CVS-v1-0-07/03/05-05:09:22 currently running on pbx (pid = 731) nection> Verbosity is at least 3 [EMAIL PROTECTED] tftpboot]# top 04:57:40 up 16 days, 23:44, 1 user, load average: 0.00, 0.00, 0.00 48 processes: 46 sleeping, 2 running, 0 zombie, 0 stopped CPU states: cpuusernice systemirq softirq iowaitidle total0.2%0.0%0.0% 0.0% 0.0%0.0% 99.8% Mem: 1025700k av, 630604k used, 395096k free, 0k shrd, 84032k buff 406804k active, 76016k inactive Swap: 2096472k av, 0k used, 2096472k free 157732k cached PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 731 root 15 0 252M 252M 3860 S 0.0 25.2 0:01 0 asterisk 734 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 743 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 744 root 15 0 252M 252M 3860 S 0.0 25.2 1:05 0 asterisk 746 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 750 root 15 0 252M 252M 3860 S 0.1 25.2 0:43 0 asterisk 760 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 763 root 15 0 252M 252M 3860 S 0.0 25.2 15:43 0 asterisk 766 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 778 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 780 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 781 root 25 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 782 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk 790 root 15 0 252M 252M 3860 S 0.0 25.2 0:00 0 asterisk What can I do to debug this? This is a production system, so during business hours I can't muck about with it, but after-hours I can pretty much do whatever I want to it... I just need some guidance. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstLinux creator to speak at Cluecon
On Wed, 20 Jul 2005, Olle E. Johansson wrote: > Brian West wrote: > > Kristian Kielhofner, the lead developer of the AstLinux project, will > > be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete > > Kristian will also be speaking at Astricon 2005 in California > http://www.astricon.net/2005/ And he'll also be speaking at Ohio Linuxfest 2005 in Columbus, Ohio http://www.ohiolinux.org ;) Kristian.. making the rounds... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stupid hold music
On Fri, 22 Jul 2005, Thomas Christie wrote: > * If you can get the song from this flash animation converted to MP3, then > it might be good (bad): > http://www.ebaumsworld.com/flash/spacepeople.html . http://damin.umlcoop.net/spacepeople.mp3 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Terry Moore-Read wrote: > I'm relatively new to Asterisk and I'm hoping attending Cluecon will be > a good way to get up to speed on the project and hear about what others > are doing with it. > > We currently use a Cisco IP phone system at my office, although I just > added an asterisk box to provide soft phones to our travelling users. > (IAX2 is a lot easier to get through firewalls than cisco's protocols). > > Terry Moore-Read > Lukins & Annis, P.S. > Spokane, WA I'm going. I'm speaking too. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Terry Moore-Read wrote: > I'm relatively new to Asterisk and I'm hoping attending Cluecon will be > a good way to get up to speed on the project and hear about what others > are doing with it. > > We currently use a Cisco IP phone system at my office, although I just > added an asterisk box to provide soft phones to our travelling users. > (IAX2 is a lot easier to get through firewalls than cisco's protocols). > > Terry Moore-Read > Lukins & Annis, P.S. > Spokane, WA I'm going. I'm speaking too. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Brian West wrote: > I'm going to be speaking about how to use valgrind, gdb and strace to > help debug issues... it can be applied to more than just asterisk. Given the following from one of my Client's boxes... pbx*CLI> show memory summary [DELETED] 7084 bytes in 435 allocations in file 'res_indications.c' 223 bytes in24 allocations in file 'chanvars.c' 51734730 bytes in 186815 allocations in file 'frame.c' 51993128 bytes allocated 188799 units total I'll be REALLY interested in your talk! Please make sure that you have take-away notes available so it doesn't evaporate into thin air after the conference! :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?
On Mon, 25 Jul 2005, Brian West wrote: > I'm going to be speaking about how to use valgrind, gdb and strace to > help debug issues... it can be applied to more than just asterisk. Given the following from one of my Client's boxes... pbx*CLI> show memory summary [DELETED] 7084 bytes in 435 allocations in file 'res_indications.c' 223 bytes in24 allocations in file 'chanvars.c' 51734730 bytes in 186815 allocations in file 'frame.c' 51993128 bytes allocated 188799 units total I'll be REALLY interested in your talk! Please make sure that you have take-away notes available so it doesn't evaporate into thin air after the conference! :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Memory Leak in Stable?
On Wed, 20 Jul 2005, Greg Boehnlein wrote: > Hello, > I have a client that has a fairly small installation (20 SIP > Phones) that is running Stable. Asterisk appears to be consuming large > quantities of memory, and growing uncontrollably to the point where after > about 6 weeks the box starts to swap itself to death. I've been keeping my > eye on it today, and in the last 12 hours, it has grown by about 8 > megabytes, and there has been no-one in the office placing any calls. Score one bug fixed for Clue Con! Mark gave me some pointers on how to debug this, and I did some legwork for him. He was able to isolate the memory leak to app_dial, and sumbmitted patches to both Stable and Head branches. Thanks to everyone that sent suggestions! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Memory Leak in Stable?
On Wed, 20 Jul 2005, Greg Boehnlein wrote: > Hello, > I have a client that has a fairly small installation (20 SIP > Phones) that is running Stable. Asterisk appears to be consuming large > quantities of memory, and growing uncontrollably to the point where after > about 6 weeks the box starts to swap itself to death. I've been keeping my > eye on it today, and in the last 12 hours, it has grown by about 8 > megabytes, and there has been no-one in the office placing any calls. Score one bug fixed for Clue Con! Mark gave me some pointers on how to debug this, and I did some legwork for him. He was able to isolate the memory leak to app_dial, and sumbmitted patches to both Stable and Head branches. Thanks to everyone that sent suggestions! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
On Thu, 11 Aug 2005, Trevor Peirce wrote: > Ing. Marlo R. Beltran G wrote: > > > I am about to buy ip pbx asterisk system but what ip phones do you > > recommend? Are polycom ip all functional with the ip pbx system??? > > We just got a Polycom IP501 for testing and have thus far been > unsuccessful at getting it to regiser with asterisk. Outgoing cals work > fine now (with authentication; verified with ethereal). Then you have something misconfigured. I have nearly 100 of them deployed at customers around the area, and they are bulletproof. Check out http://www.krisk.org/asterisk/pcom/ipmid.cfg for an example. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
On Fri, 12 Aug 2005, Matt Florell wrote: > Short answer: NO > > Long answer: you have to send it to Digium for them to do an upgrade, > they don't have an official process for this yet and won't give you a > price, I have called and asked them many times. They also mention > upgrades from your 405/410 to a 406/411 are available too, but again > no specifics. Supposedly if you have a card with the 2nd gen firmware > on it you can upgrade to the third gen firmware, whenever it would > come out, in the field. Hmm.. That's funny. I called yesterday and talked to someone who told me the upgrade to Second Gen firmware was free, but that if I wanted to add the Echo Cancelling module, it would be $850. Since I do not have any major echo issues that software echo cancelling can't fix, I declined the upgrade. They even offered me an advance replacement option as long as I provided a Credit Card. The RMA process was painless. I spoke with Joy Lister. I should have my new card early next week. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] yet another Asterisk and VMware question
On Fri, 12 Aug 2005, Bruce Leetch wrote: > Am I banging my head against at Windows/VMware/Linux/Asterisk > incompatibility? Or can this work and I'm just doing something stupid > (always a possibility with me). It's not going to work. Vmware presents a complete Virtual PC, so unless EMC / Vmware release drivers to specifically connect the Virtual PC to the real PC hardware, you are in trouble. The do a pass-through w/ USB, Serial and Paralell, but those are a different story. > With a great amount of effort, I can drum up a spare machine, but I REALLY > don't want to do this and would much prefer the VMware setup. Any advice > will be welcomed. I'm afraid that under any Virtual platform (CoLinux, Vmware, MS Virutal PC) you are SOL as far as real access to hardware on the host PCI bus w/out special drivers written specifically for that purpose. On the other hand, I'm sure that Vmware would be happy to help you out if you gave them a couple of million bucks! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] yet another Asterisk and VMware question
On Fri, 12 Aug 2005, Tom Rymes wrote: > > VMWare is a virtual machine and has nothing to do with the physical > > layout of the box (which is why you can migrate vmware images > > across machines for example). > > > > If you want to run Asterisk under Linux setup a box to run it. > > Agreed. You would be better to grab a used $200 machine and install > linux & Asterisk on it. Unless you are scaling up to at least 10+ > simultaneous calls, I would imagine that something you have lying > around would handle it. > > If you insist on VMWare, I would imagine that you could configure a > Sipura SPA-3000 to provide incoming (FXO) and analog extension (FXS) > ports This works. I've done it on occasion for testing. However, because virtual PCs rarely operate on a real-time clock, mostly emulating these features, you will find that anything that read/writes to disk will suck badly. For example, it is nearly impossible to use the Voicemail features of Asterisk under Vmware, CoLinux or UserMode Linux. Believe me, I've tried! ;) This is one of the main reasons that AstWind has stagnated. The timing granularity of the virtual machines is not acceptable for doing anything IO related. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ipVolution
On Wed, 28 Dec 2005, Goran Skular wrote: > Hi, > > Anybody have some experience and did some testing with ipVolution E1/T1 > cards? I chalked these up to VaporWare... -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Bugs that Need Your Input!
Hello, I know that Mog was trying to get the bug-tracker cleaned up as the number of bugs has increased substantially over the past few months. I figured that I would do my part to bring attention to a couple of bugs that are interesting and have some wide reaching impact. That being said: http://bugs.digium.com/view.php?id=5970 - This has a patch created by Corydon, but it has segfaulted on three different boxes. Backtraces are attached to help track this down. http://bugs.digium.com/view.php?id=6027 - This is related to the bug above and has a different patch that might be relevant. http://bugs.digium.com/view.php?id=5374 - This is a patch that allows for the asychronous generation of outgoing frames. Very cool. Needs a decision by the Core Developers (Read Kram / KPFleming). BJ has been keeping this in synch w/ Trunk in his own branch. http://bugs.digium.com/view.php?id=5090 - T.38 Support anyone? http://bugs.digium.com/view.php?id=5574 - A nifty new app called Find Me, Follow Me that does exactly what it says. If you can, take a look at these and add your entropy to the mix! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] O'Reilly's Etel Conference
Hey there, Just wanted to drop a line and let people know that I'll be heading to San Francisco for O'Reilly's Etel. If you are interested in attending, there are some free passes floating around. If anyone is interested in getting together for a beer, let me know! Info on the conference can be found here: http://conferences.oreillynet.com/etel2006/ I'm looking forward to an ejoyable uber-geek experience! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] 1.0.10 to 1.2.1 upgrade..is it worth it?
On Wed, 11 Jan 2006, stevanus wrote: > Hi, > > As I've dealt with asterisk 1.0.10 successfully, I wonder what the > benefit I will get from upgrading to 1.2.1.. > [Of course I know there're lot of new interesting stuffs in 1.2.1, but > are they stable already?] > > Does the 1.2.1 need more resources, more power hungry? > > Anyone has success story with asterisk 1.2.1 please share :) > > Thank you... If it is of any assistance, we just dropped a 90 seat call center with a predictive dialer onto an Asterisk 1.2 box this afternoon w/ a TE405P in it. Over the course of 1/2 hour, they sent 3,000 calls through the box and it is still going strong! ;) It's being hammered without mercy, and this is a single Intel 3Ghz CPU doing SIP -> ZAP w/ G729 and SIP -> SIP Passthrough. They are averaging about a call a second through the box. We had more problems with our SIP Termination upstream not being able to handle the load... Asterisk didn't blink. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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: DTMF Issues With Asterisk 1.2 IVR
On Thu, 12 Jan 2006, Steven wrote: > If it is a toll free number, it may be related to > http://bugs.digium.com/view.php?id=5266 . You may also want to investigate: http://bugs.digium.com/view.php?id=5970 as well as: http://bugs.digium.com/view.php?id=6027 There are several people that are putting some effort into these issues, and if we can get some input from a wider audience it might help us squash the outstanding issues faster. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] double ringing tone on asterisk 1.2
On Sat, 14 Jan 2006, Brian Capouch wrote: > Rich Adamson wrote: > > > > > > Since there does not seem to be anyone else complaining about the same > > problem, there must be something in your config that is causing it. > > Without specific copy/paste samples of what you've configured, no one > > is going to be able to guess at what you are doing. > > > > Given the issue is happening with both PRI's and IAX links, I'd have to > > guess that you've got something wrong in extensions.conf. > > > > Actually, I'm not complaining, but I've experienced a similar problem. > For me it only happens with IAX calls, and from there only IAX calls > with a particular ITSP. I have that problem when using IAX transport as well. I've just ignored it until now, cause it is on a home box. > But his description rings very true: there are two rings per time that > the far-end phone is ringing, and just like he said, one of them sounds > genuine and the other sounds like perhaps my provider is adding it to > the audio coming my way. > > The dinger is that it doesn't happen on all calls; it's more or less > random, but frequent. > > I don't mind it, though, so I'd never complained about it on the list. > > FWIW. > > B. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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: 1.2.1 "Silence suppression is disabled" whatthehell?
On Mon, 16 Jan 2006, Koopmann, Jan-Peter wrote: > On Sunday, January 15, 2006 12:21 AM Tony Mountifield wrote: > > > In article <[EMAIL PROTECTED]>, > > Pisac <[EMAIL PROTECTED]> wrote: > >> I've found something here: http://bugs.digium.com/view.php?id=5374 > >> > >> but I don't understand how this can be connected to my problem :-( > > > > It looks like the maintainer of the BRIstuff distribution might have > > decided that patch was worth including, even though it is not in the > > standard 1.2.1. That does give scope for confusion though! > > Look at the CHANGES. I was the one who convinced kapjeod to put that patch in > the current bristuff distribution. So yes: It is in bristuff as of 1F: > > 0.3.0-PRE-1f > - THIS IS GETTING CLOSER TO A STABLE RELEASE, USE IN PRODUCTION AT YOUR > OWN RISK! > - merged patch for bug 5697 (meetme) > - merged patch for bug 5374 (asynchronous generation of outgoing frames) > - _finally_ fixed "sending-nonRFCcompliant-SIP-NOTIFYs" bug (asterisk, > extension states) > - some debug output clean ups in libpri Per other discussions on this issue, that patch breaks a ton of stuff (IAX Jitter Buffer, Music On Hold etc..). Although it does solve some issues, it really needs to be reviewed by "They Who Decide Such Things" as it has far reaching impact on other areas of the system. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] MoH trouble with latest bristuff (0.3.0-PRE-1f)
On Tue, 17 Jan 2006, Kib Eki wrote: > Hi Karsten, > > I have the same problem. MusicOnHold sounds awful. The PRE-1e does not have > this > problem. I have two identical systems (hard-/software). One system has the > problem the other does not. I thought i could be timing problem or interrupt > conflict. But we could not find out the problem. > > Bernd > > Karsten Wemheuer wrote: > > Hi, > > > > I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When > > I activate music-on-hold on a SIP-to-SIP connection, the music sounds > > like in a fast-forward play mode. On the *-console I can see much lines > > like this: > > -- Silence suppression is disabled (option_silence_suppression=0 > > chan->timingfd=18) > > > > What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but > > there was another issue, so I have to upgrade). bristuff 3.0 apparently contains the "asynchronous frame generator patch" from http://bugs.digium.com/view.php?id=5374 This apparently breaks a LOT of other stuff, MOH being one of them. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 down because of cdr
On Tue, 17 Jan 2006, Jean-Michel Hiver wrote: > Dov Bigio a ?crit : > > >Ok.. but I don't use Real Time at all. > >I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages > >or at least just logged, but without stopping. > > > > > What would be even nicer would be for * to buffer it for a while before > it starts dropping cdrs... Might want to use cdr_odbc instead. The issue that you are discussing was brought up by BKW last year, and I was under the impression that it had been patched. I.E. that when MySQL died, Asterisk would deadlock. I'm pretty sure that cdr_odbc (which is cake configure) handles this situation properly. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 down because of cdr
On Tue, 17 Jan 2006, Alexander Lopez wrote: > Buffer! For how long? How big of a buffer? If I can buffer 10-20 calls >that might work if I have a light use PBX but 100-2000 buffered calls >may not hold a busy PBX. OK so make it configurable, With any luck you >won't know how much to put so you will allocate more than you need, >using more memory for a single senario. > > My solution, make sure your DB is stable. I would rather put my effort >in building a better solution than counting on insurance to bail me out. Hehehe.. it should be a dynamic buffer that should grow / shrink as neccessary and consume as much memory as it needs until the database backend comes online. It should do periodic checks of the server to see if it is available and then spool the buffered calls out to it, but only spool stuff in the event the connection is not available. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 down because of cdr
On Tue, 17 Jan 2006 [EMAIL PROTECTED] wrote: > Buffers don't have to be in memory. My suggestion on the solution would be > to buffer the CDR info into a backup file based database (configurable > filename/path) on the local filesystem (or NFS mounted system for > redundancy) and then when the SQL database connection is restored then it > spends a second dumping the buffered CDR info from the file into the > database and erases the file (or empties it). Why not just use the astdb instead of a file? Dundi does it.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Dialstatus Oddity in 1.2
Hello all,
I am working on a creating some intelligent failover dial-plan
logic and I'm running into something that I'd like some feedback on.
Basically, it appears that if you place a call to an IAX2 peer that
refuses the connection, or is unavailable, a NOANSWER dialstatus is
returned.
Example:
-- Executing Macro("IAX2/cubix-19", "nocdial|IAX2/[EMAIL
PROTECTED]/1216410") in new stack
-- Executing Dial("IAX2/cubix-19", "IAX2/[EMAIL PROTECTED]/1216410|30")
in new stack
-- Called [EMAIL PROTECTED]/1216410
Jan 21 19:16:07 WARNING[1114]: chan_iax2.c:6970 socket_read: Call rejected by
207.166.192.188: No authority found
-- Hungup 'IAX2/pbx1-21'
== No one is available to answer at this time (1:0/0/0)
-- Executing Goto("IAX2/cubix-19", "s-NOANSWER|1") in new stack
-- Goto (macro-nocdial,s-NOANSWER,1)
-- Executing Hangup("IAX2/cubix-19", "") in new stack
-- Hungup 'IAX2/cubix-19'
Shouldn't that return CONGESTION instead? I thought that NOANSWER was
reserved for calls that reach app_dial's timeout limit?
Or am I just missing something simple?
Here is the relevant extensions.conf logic that I am using
[e164]
; Dundi
exten => _1NXXNXX,1,Macro(dundi-e164,${EXTEN})
; Dispatch First Trunk
exten => _1NXXNXX,2,Macro(nocdial,${TRUNK}/${EXTEN})
exten => _1NXXNXX,3,ResetCDR
; On Failure, Dispatch Second Trunk
exten => _1NXXNXX,4,Macro(nocdial,${TRUNK2}/${EXTEN})
exten => _1NXXNXX,5,ResetCDR
; Third time is a charm?
exten => _1NXXNXX,6,Macro(nocdial,${TRUNK3}/${EXTEN})
[macro-nocdial]
exten => s,1,Dial(${ARG1},30)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(15)
exten => s-BUSY,2,Hangup
exten => s-CONGESTION,1,NoOp
exten => s-CHANUNAVAIL,1,NoOp
exten => s-.,1,Goto(s-NOANSWER,1)
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
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Re: [Asterisk-Users] T3 Mux and Asterisk Question
On Sun, 22 Jan 2006, Steve Totaro wrote: > I have a T3 coming from my carrier. From there I want to use an Adtran > mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad > T1/PRI equipped servers. > > Everything seems very straight forward with the exception of the D > channels for the T1/PRI. > > I am not very familiar with large circuits such as T3s. I know that I > can use one D channel per set of quad port on each server. So if each > server has a quad port card, I can use one channel as the D channel for > all four spans. > > That gives me seven D channels in my setup. Does anyone know how the > Mux handles these D channels onto the T3? My guess is the Mux is simple > going to send all of the channels onto the T3 without modifying > anything. That's correct. The T1 spans on the DS3 are completely independent of the clocking on the DS3. The D-channel and timing is something that will be handled by your upstream Telco and the switch that you'll be connecting to. Or, your own box.. ;) > What I would really like to do is have one D channel coming in on the T3 > and have it split between each of the T1/PRI or even better one D > channel per quad (I know Asterisk can do that). > > Is it possible? No. > If the Adtran mx2800 cannot do it, is there anther > product that can. I have looked at the RAD Optimux T3 product but have > had great experience with Adtran products. The price is the same but > the Adtran allows for two controller cards so it seems to have more > built in redundancy. > > Any tips would be appreciated. Adtran's MX-2800 is our choice for Muxes. They are solid, reliable and work well. Adtran's technical support is amazing. When you purchase an MX-2800, you are immediately given access to the Adtran Carrier support group, which doesn't even blink about sending out an advance replacement unit overnight if you ask. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] T3 Mux and Asterisk Question
On Sun, 22 Jan 2006, Steve Totaro wrote: > Thanks for some answers, that is what I thought. > > Asterisk is NFAS capable so I am looking at seven D channels on the T3 I > guess. I don't want to put a D channel on each T1 or I will lose > several channels that could be used for calls. > > I wonder if there is any way that Asterisk can do NFAS across multiple > servers. I would put two cards in a box but they will be doing g729 > transcoding so I don't want to push it, so it is one per server. Hehehe.. Ask your Telco if they can provision E1 for you. ;) The Digium cards can handle E1 or T1, and if you go E1 you'll get 30 channels instead of 24 on the span. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] T3 Mux and Asterisk Question
On Mon, 23 Jan 2006, Kevin P. Fleming wrote: > Greg Boehnlein wrote: > > (Steve Totaro wrote:) > > >>What I would really like to do is have one D channel coming in on the T3 > >>and have it split between each of the T1/PRI or even better one D > >>channel per quad (I know Asterisk can do that). > >> > >>Is it possible? > > > > > > No. > > Actually, it is, using an Adtran Atlas with a DS3 interface and DS1 > interfaces. Not cheap, but possible. Yeah, but he's already stated that he will be using MX2800 muxes. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] yet another Asterisk and VMware question
On Sat, 13 Aug 2005, Lull, Rick wrote: > This is one of the main reasons that AstWind has stagnated. The timing > granularity of the virtual machines is not acceptable for doing anything > > IO related. > > Just since I am curious, what version of VMWare did you use and what kind of > box where you running on? 4.5 and 5.0 variants of Vmware Workstation and CoLinux on top of a Windows host environment. Same applies for running Vmware under a Linux host environment as well. Machines where 2.8 Ghz Dell boxes w/ 1.5 gigs of ram. > I've just moved my * box to a VM on ESX server and didn't play with > voicemail until you mentioned it - now Allison's voice cuts in and out. > Sounds like I am going to have to go back to the box I was running on > previously. My original box is a P3 500 desktop while my VMWare ESX box is a > dual P3 1.4GHz HP Proliant server. You would probably get the BEST performance by using User Mode Linux, w/ the latest SKAS patches. I know someone that does this and only occasionally has a dropped frame in audio here and there. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Cop as a firewall and QOS
On Thu, 18 Aug 2005, Holden Hao wrote: > > I don't mind buying an appliance to get something solid but IP Cop just > > looks better than he appliances I see out there. > > Astaro has been getting good reviews from Linux World. They have an > appliance solution or a self-install solution. It features: > > -Firewall > -VPN Gateway > -Intrusion Protection > -SPAM Filtering > -Anti-Virus > -Management Platform > -Surf and Spyware Protection > > The details of the features are impressive. For the details visit: > > http://www.astaro.com > > You can download a 30-day demo. > > If cost will be a problem, IP Cop is also a good solution. This is > what we have been using. I second the vote on Astaro. They have, without a doubt, fused the best features of a Perimiter Security Device w/ Open Software and an excellent GUI. Version 6.0 now includes a SIP proxy as well! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
On Thu, 18 Aug 2005, Hadar Pedhazur wrote: > First, many thanks to Greg Boehnlein for his patch to chan_agent.c > for adding a "preackannounce" option. > > I am running CVS HEAD from 2005/07/31, and the patch failed in a > few hunks, since the code was refactored to add in some CASE > statements where there were compound if statements before. > > Anyway, I have successfully updated the patch to work against head > as of 3 weeks ago, and would happily share that with anyone who is > interested (just drop me a line off list). > > If a "diff" is preferable to the full 70k of "C", just let me know > what the correct options are for creating a diff suitable for > patching the asterisk tree. > > OK, that said, I have a few questions and comments on this topic. > This is my first use of the Queue command (very successfully so > far), but I am afraid that expanding my use will require further > patches, and I would like to verify that first. > > 1) If I use the syntax: > > Member => SIP/100 (rather than member => Agent/100, which maps to > SIP/100) > > Then "ackcall" isn't used at all. In other words, a "hard-wired" > member seems to ignore the agents.conf file completely. Is this > the desired behavior? (It isn't for me...) It is the correct behavior because when you use SIP/100, chan_sip (which has no concept of an agent) is being used instead of chan_agent to deliver the call. Think of chan_agent as an intermediary between the PBX core and the physical endpoint. chan_agent accepts the call, puts the caller on hold and then grabs chan_sip to complete the other side of the call. When all requirements of chan_agent are met, it then proceeds to bridge the two calls together and get out of the way. > 2) Since agents.conf is a separate file from queues.conf, having > multiple queues does _not_ permit multiple "preackannounce" > messages, each tied to a different queue (this strikes me as > having better been patched into the Queue command). Similarly, you > can't have one queue that has "ackcall=yes", and another with > "ackcall=no". Right. It's a botched design and chan_agent's design doesn't lend itself to being very helpful in the process, but that is where it had to go. This is the reason that I dropped work on it, as ICD was a much more intelligent design at the time. > 3) I have the _exact_ same source version of CVS HEAD (from > 2005/07/31) running on different servers (after a "cvs co", I tar > the source so that I can be sure I'm running _identical_ > versions). > > On one machine, when an Agent logs in, I can see it in the DB, > "database show" shows a key of: > > //Agents/1001 : [EMAIL PROTECTED];1001 > > On another machine, the DB shows _nothing_, yet the > AgentCallbackLogin application works correctly (logging agents in > and out), and shows the correct mapping on the CLI during a login. > Still, the DB has _no trace_ of the Agents. I can't explain the > difference in behavior, and would _love_ to have someone solve > that mystery for me. > > I'm hoping that I am missing something obvious in the interaction > between the Queue command and the Agents channel, and that some > kind soul here will educate me. Otherwise, I think I might be off > to doing more work in "C" than I ever though I would again in my > life ;-). Don't have an answer for this one. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
On Fri, 26 Aug 2005, Hadar Pedhazur wrote: > Hmmm. I am often surprised when I don't get a response to a post that I > think would interest at least _one_ person in the community. This one > surprised me a little more, since I offered some code ;-). > > This morning, I just got a bounce notice that it was undelivered, which > might explain it, except that I received the original post back through > the list, so I don't understand it at all... > > Anyway, I solved the one "bone-headed" problem that I describe below, > namely why did the agents show up in one DB and not the other. I didn't > set the "persistent" keyword in the agents.conf file (doh...). > > All of my other questions still apply, as well as my offer to share the > code/patch. I just got your message and responded to it! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Reboot Script
On Mon, 29 Aug 2005, Kristian Kielhofner wrote: > Matt, > > It sure is! You should be testing it! :) Test it and see, but 1.2 > will be "STABLE" pretty soon here... No. No NO NO NO NO! :) 1.2 will never be called Stable, based on the controversy surrounding the naming moniker. Do you ever heard Linux 2.0, 2.2 or 2.4 referred to as stable? No. People tend to confuse "Stable" as in "No Additional Features, Bug Fixes Only" with "Stable" as in "It never has any problems". But.. I may be too late on this one.. ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 1.6.7 Firmware Messages Button
Hello, I recently updated some Polycom 501 phones to the new 1.6.7 firmware, and have lost the ability to do "One Touch" voicemail access via the messages button. I've verified that I have the correct XML tags set in the phone config, I.E.: msg.bypassInstantMessage="1" mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="85100" I've wiped the phone clean, and re-installed firmware and configs, and it still acts as if the msg.bypassInstantMessage tag is set to 0 and displays the status of the messages in the mailbox. I didn't see anything in the release notes indicating a change in the behavior of these tags. Anyone have any suggestions? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
On Sun, 30 Jul 2006, Peter Johnson wrote: > How about up.oneTouchVoiceMail="1" in your sip.cfg > > Peter Ahhh... that tag wasn't in my config generator script, so I must have set it by hand in the old ones. That does the trick! I owe you a beer! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] VoipNow 1.2.0 Beta
On Sat, 29 Jul 2006, Tom Vile wrote: > Did you look on the site? > > http://www.4psa.com/products/voipnow/demo.php Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESK user API. NEXT! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
On Sat, 29 Jul 2006, Douglas Garstang wrote: > You have a config generator script for the Polycom XML files? What did you > build that with? Bash scripts and some sed logic. It isn't pretty, but it works. # This creates the actual SED script that we use to modify the template echo "s/reg.1.displayName=\"\"/reg.1.displayName=\"$NAME\"/" >> $$.sed echo "s/reg.1.address=\"\"/reg.1.address=\"$EXTEN\"/" >> $$.sed echo "s/reg.1.label=\"\"/reg.1.label=\"$EXTEN\"/" >> $$.sed echo "s/reg.1.auth.userId=\"\"/reg.1.auth.userId=\"$EXTEN\"/" >> $$.sed echo "s/reg.1.auth.password=\"\"/reg.1.auth.password=\"$EXTEN\"/" >> $$.sed echo "s/reg.1.server.1.address=\"\"/reg.1.server.1.address=\"$SERVER\"/" >> $$.sed echo "s/reg.1.server.1.register=\"\"/reg.1.server.1.register=\"1\"/" >> $$.sed echo "s/CONFIG_FILES=\"phone1.cfg, sip.cfg\"/CONFIG_FILES=\"phone$EXTEN.cfg, sip-n2net.cfg\"/" >> $$.sed echo "s/LOG_FILE_DIRECTORY=\"\"/LOG_FILE_DIRECTORY=\"$EXTEN\"/" >> $$.sed echo "s/OVERRIDES_DIRECTORY=\"\"/OVERRIDES_DIRECTORY=\"$EXTEN\"/" >> $$.sed echo "s/CONTACTS_DIRECTORY=\"\"/CONTACTS_DIRECTORY=\"$EXTEN\"/" >> $$.sed # Now, we copy the templates to their new filenames cp phone1.cfg phone$EXTEN.cfg cp .cfg $MAC.cfg # Sed it up sed -i -f $$.sed phone$EXTEN.cfg sed -i -f $$.sed $MAC.cfg # Create Directories and Set Perms mkdir $EXTEN chown -R PlcmSpIp:PlcmSpIp * chmod -R a+rwx * # Voila! rm -f $$.sed phone$EXTEN.cfg~ $MAC.cfg~ -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
On Fri, 4 Aug 2006, Steve Totaro wrote: > > If your M13 is coming up clean, I'd double check the continuity to > > those ports from the DSX connections out to the patch panel. > > That was exactly the issue. The amphenol cable was loose on one end. > What is the best way to fasten these things? There is only a screw on > one side. I am duct taping these things down for now but what is the > standard way of fastening them? The Adtran didn't come with a clip or > velcro or anything, neither did the TenorAX boxes I am using. We went the Duct Tape route initially for the side that doesn't have the screw, but eventually moved to some velcro contraption that one of our techs dreamed up while drinking at the bar downstairs. So far, so good. We just stay far away from the connections on the back of the rack! ;) The MX200 is a fantastic product, with great support! I'll never even consider buying a mux from another company after the support that Adtran has given us with ours. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] VoipNow 1.2.0 Beta
On Tue, 8 Aug 2006, mitcheloc wrote: > > > > On Sat, 29 Jul 2006, Tom Vile wrote: > > > > > Did you look on the site? > > > > > > http://www.4psa.com/products/voipnow/demo.php > > > > Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESK > > user API. NEXT! > > Instead of flaming, you could accept that not everyone makes software with > just you in mind. It isn't a flame. It is an observation and an opinion. I'm happy that people are creating new interfaces for Asterisk management, but that doesn't mean that I have to like them or the technology that they are based on. People choose to do all sorts of silly things, some more silly than others. PSA knows PLESK, so they have used it to build their UI. I use PLESK for my Web hosting customers, and although it works, it certainly is not pretty nor is it something that I would reccomend as a platform for building an Asterisk management application. Your mileage may vary. Use what works for you. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 VoipNow 1.2.0 Beta
On Tue, 8 Aug 2006, Matthew Warren wrote: > Yes it is an addon of Plesk, thats stating the obvious. But while your > complaining about people writing stuff to use what are you doing. If your > not a developer don't critisize the developers. I see nothing more than you > displaying that you are the Vice President of a 2 man consulting firm. > Which means you have to sell other peoples developed products. Wow. Such hostility. Some corrections. 1. I am using a variety of customer built and off the shelf products, both stuff that I've written, commissioned to have written and/or purchased from companies such as PBXware, Switchvox, VoiceRoute etc. My customer's needs drive my decisions as to what should be deployed, not a religious fervor. However, my customers also pay me to make solid decisions on technology based on my (considerable) experience in the field. 2. N2Net, of which I an the Vice President, started 11 years ago as a 1 man consulting company. We are quite a bit larger now, but that isn't important. What is important is that you haven't even taken the time to verify who you are talking to, and/or what their credentials are. Bad move on your part, and probably not the smartest thing to fire off to a list such as this. I'll chalk it up to a mistake on your part. A google search of your name and the term Asterisk doesn't show a whole lot. You might consider doing the same with my name "Gregory Boehnlein Asterisk" to see what comes up. > Not to mention you are being critical of plesk, yet you use to host you > websites for your business. Yes.. I host several thousand websites with PLESK, and as such am quite well positioned to voice an opinion about their software and methodologies. As far as Web control panels go, they are the best in the industry, but the industry has pretty low standards. I dislike PLESK for a variety of reasons and personally see that someone using it as a basis for a Web UI for Asterisk is trying to push a square peg through a round hole. That doesn't mean that VoipNow sucks, just means that I'll probably not use it or reccomend it to my clients because I think the architecture and design are fudamentally flawed, and it's my reputation on the line. However, if someone asked me for a reccomendation on a Control Panel software for Web Hosting, I would have no hesitations reccomending PLESK to them, provided they understand the limitations of the platform. > Dude, we all have opinions, like crapholes they > all stink, your's just stood out. Everyone is entitled to their own opinion, and you have every right to have your own. I don't agree with it, but I'm not going to resort to childish insults and petty comments. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] High Availability with PRI failover
On Fri, 11 Aug 2006, Senad Jordanovic wrote: > [EMAIL PROTECTED] wrote: > > Hi > > > > After a month or so using Asterisk we've had or first downtime period > > due to a faulty RAM chip on the server, so we're starting to think > > about the possible high-availability solutions. > > Hi > > If you can afford it, below will give you total fault tolerant solution. > > http://www.bicomsystems.com/products/C/P/319/255_2797/ I think I can afford that. When I go to the link I get: Under Maintenance Please come back later. Bicomsystems.com ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 Performance without RTP?
On Sat, 26 Aug 2006, Kelvin Williams wrote: > If Asterisk was used to set up and tear down calls, and using canreinvite > allowing the RTP to pass from end-point to end-point, how many calls could > Asterisk handle at once? I've pushed over 1,000 concurrent calls this way using the SIPP program for SIP performance testing. There was some tuning that needed to be done, but it worked. Never went that far in production, though. > I ask because I have been utilizing OpenSER but find myself constantly > needing Asterisk to do this or that, and would like to move OpenSER into > more of a Registration server, and letting Asterisk handle all of my calls I > understand that the set up and tear down may be a tad slower, but > programming (using AGI, etc.) would definitely outweigh the timing IMO. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 Performance without RTP?
On Mon, 28 Aug 2006, Andrew Kohlsmith wrote: > On Monday 28 August 2006 13:02, Greg Boehnlein wrote: > > I've pushed over 1,000 concurrent calls this way using the SIPP program > > for SIP performance testing. There was some tuning that needed to be done, > > but it worked. Never went that far in production, though. > > May you share some of your tuning with us? What gotchas did you discover? Just making sure your dial-plan as efficient as possible, that you have enough sockets and open file limits in the kernel, not connecting to the CLI console, never, ever using cdr_mysql or cdr_odbc for your CDR records (locking / contention issues) etc... Lots of basic common sense stuff that you often forget about.. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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 Performance without RTP?
On Tue, 29 Aug 2006, Nick Hoffman wrote: > On Tue August 29 2006 04:39, Greg Boehnlein <[EMAIL PROTECTED]> wrote: > > On Mon, 28 Aug 2006, Andrew Kohlsmith wrote: > > > On Monday 28 August 2006 13:02, Greg Boehnlein wrote: > > > > I've pushed over 1,000 concurrent calls this way using the SIPP > > > > program for SIP performance testing. There was some tuning that > > > > needed to be done, but it worked. Never went that far in production, > > > > though. > > > > > > May you share some of your tuning with us? What gotchas did you > > > discover? > > > > Just making sure your dial-plan as efficient as possible, that you have > > enough sockets and open file limits in the kernel, not connecting to the > > CLI console, never, ever using cdr_mysql or cdr_odbc for your CDR > > records (locking / contention issues) etc... > > > > Lots of basic common sense stuff that you often forget about.. :) > > Hi Greg. What problems/performance issues does cdr_mysql introduce? If the database is unavailable, or performance is slow, it can cause a blocking condition that will stop the entire system from processing anything. It may have been fixed since then, but I thought that cdr_mysql was deprecated.. > -- Nick > e: [EMAIL PROTECTED] > p: +61 7 5591 3588 > f: +61 7 5591 6588 > > If you receive this email by mistake, please notify us and do not make any > use of the email. We do not waive any privilege, confidentiality or > copyright associated with it. > > -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF between cisco and sipura going through asterisk
On Tue, 29 Aug 2006, Benjamin Lawetz wrote: > Hello all, > > we're having an issue with DTMFs being sent to Sipura's. Calls are > originating from a Cisco AS5300 being sent to asterisk which in turn sends > it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows > the same problem with a cheap answering machine). The DTMFs sent from the > AS5300 aren't recognised by the legacy PBX. > > - DTMFs are recognised correctly on the asterisk (when we check voicemail) > - The cisco is setup with dtmf-relay rtp-nte > - in sip.conf the cisco and sipura are set to rfc2833 > > If I set the cisco in dtmf-relay rtp-cisco it works on the sipura, but not > on the asterisk. > > Unfortunately I can only set one dtmf-relay mode on the cisco. Is there > anything I can change on asterisk or sipura to get the sipura to work with > the rtp-nte (or to get asterisk to work with the cisco-rtp)? > > Any hints can help, Ben, What version of Aserisk are you using? If it is the 1.2 series, there are all sorts of RFC-2833 DTMF Relay issues that can crop up. My suggestion is that if you are willing to take the time, it might be worth it to Upgrade to the pre-release version of Asterisk that is currently in TRUNK. This supports the new Variable Length DTMF code that should knock out nearly all of the DTMF issues that Asterisk has had. The 1.2 and earlier RTP stack and RFC-2833 implementation, while not technically wrong according to the RFC, did things a bit differently than the rest of the world has chosen, and therefore can cause DTMF instability. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] SIPP problem
On Sat, 2 Sep 2006, Diego Quintana Cruz wrote: > Hi everybody, > I'm trying to load-test my Asterisk PBX using SIPP, but I always > getting errors, I followed the instructions given in [1] which mainly > was to create the user sipp in sip.conf and the dialing plan for his > context in extensions.conf > > I'm using Asterisk 1.0.10 > > Any ideas or tutorial on how using SIP? Here are my notes on the subject: http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Digium at Ohio Linuxfest
Anyone heading out to Ohio Linuxfest? Digium is going to have a booth there. I would appreciate people checking this out and digging it if they feel it would be appropriate. Nothing like a FREE Open Source conference with live penguins! :) http://digg.com/linux_unix/Ohio_Linux_Fest_2006 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] TDM2420E Availaibility
Hello, Rumor has it that the TDM2400 series cards will be available in the next week or so. If you are a distributor that has pricing / availability information, please contact me offlist. I am putting together a solution for a client that will require a TDM2420E (8 Port FXS w/ Echo Can) and I need to compare it against an Audiocodes 8 port SIP <-> FXS gateway. Thanks -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual PRI fail over
On Tue, 11 Oct 2005, Tom wrote: > > > I currently have a single PRI however we are getting a second PRI, and the > provider (qwest) wants to know if our PBX supports GSAS (they say its a > redundant d-channel technology but searching on google for GSAS reveals less > than nothing). I've set something similar up before on a cisco 5350, where if > one of the PRIs fails, all of the calls destined for either PRI will be routed > down the one that didn't fail. Basically the 2 PRIs are bonded together, and > act as one. During normal operation the calls come down each PRI in a load > balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on > the other). > > Is there any way to set something similar to this up in Asterisk? > Tom They are probably talking about NFAS. And after looking at the date of this message, it's probably a moot point. But I'm not sure if NFAS is supported under Libpri/Zaptel. I'd suspect that if it is, it might not be widely deployed and/or tested with it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone know who is in this picture?
On Wed, 2 Nov 2005, Matt Darnell wrote: > Well that didn't take long! > > He was a really nice guyI bet it would be a blast to go have a beer with > him. > > We met him at the Internet Telephony Expo. Read his bio on Rotten.Com. I'm surprised to see him posing with Women. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone know who is in this picture?
On Fri, 18 Nov 2005 [EMAIL PROTECTED] wrote: > I couldn't find his bio on rotten.com http://www.rotten.com/library/bio/hackers/captain-crunch/ -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_icd anyone? on 1.2?
On Tue, 22 Nov 2005, Lenz wrote: > I also have never found anybody running an Asterisk system using app_icd. > Maybe app_queue is now after all flexible enough to be used in most cases. > Anybody else using different apps for Asterisk call centre applications? I suspect that since the authors of ICD are no longer really submitting patches to Asterisk, that ICD for Asterisk is probably end of life, unless someone wants to pick up and take on maintenance of the code. As far as I know, ICD was never officially accepted into the mainline tree and was always kept as an outside project. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] v1-2 install mkdep loop
On Mon, 21 Nov 2005, Bob Knight wrote: > Just pulled a v1-2 onto a system that was running a v1-0. > > Zaptel and libpri, build and install just fine. > Building asterisk is fine. > But when I try to do a make install on asterisk, it goes into an > infinite loop doing on .depend doing: build_tools/mkdep > > I did the same thing on another box the other day with a different pull > and did not have any problems. Do you think this is something related > to this box? Hi Bob! Long live the PM3! This is an issue that many many people have been running into, and has been discussed on the dev list. Check the following: http://lists.digium.com/pipermail/asterisk-dev/2005-November/016509.html I'm not sure there is a specific fix, although there are many suggestions in that thread. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
