Re: [asterisk-users] Asterisk Trunk and normal
Usually you'd only need to go to the trunk to get features that haven't made it into the "stable" tarballs yet. On Tue, Sep 2, 2008 at 10:37 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote: > Yes I mean the trunk for the development, when I have to select such version > and when I can use the normal? > > Regards > Bilal > > > --- On Tue, 9/2/08, Erik Anderson <[EMAIL PROTECTED]> wrote: > >> From: Erik Anderson <[EMAIL PROTECTED]> >> Subject: Re: [asterisk-users] Asterisk Trunk and normal >> To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial >> Discussion" >> Date: Tuesday, September 2, 2008, 11:33 AM >> Bilal - I think you're perhaps confusing two meanings of >> the word >> "trunk". In this case, "trunk" is >> referring to the trunk of the SVN >> development repository, not SIP or IAX trunks. This can be >> seen as the >> main development area for asterisk. >> >> On Tue, Sep 2, 2008 at 10:22 AM, bilal ghayyad >> <[EMAIL PROTECTED]> wrote: >> > Sorry, but I did not find in the below link anything >> answering the difference between the trunk and not trunk >> version? When to use asterisk trunk and asterisk normal? >> > >> > Regards >> > Bilal >> > >> > >> > --- On Tue, 9/2/08, Dan Julius >> <[EMAIL PROTECTED]> wrote: >> > >> >> From: Dan Julius <[EMAIL PROTECTED]> >> >> Subject: Re: [asterisk-users] Asterisk Trunk and >> normal >> >> To: [EMAIL PROTECTED], "Asterisk Users >> Mailing List - Non-Commercial Discussion" >> >> >> Date: Tuesday, September 2, 2008, 9:33 AM >> >> Hi, >> >> >> >> checkout >> >> >> http://svnbook.red-bean.com/en/1.4/svn.tour.importing.html#svn.tour.importing.layout >> >> this explains about versioning >> >> >> >> Dan >> >> >> >> On Tue, Sep 2, 2008 at 3:56 PM, bilal ghayyad >> >> <[EMAIL PROTECTED]> wrote: >> >> >> >> > Hi List; >> >> > >> >> > I see and hear about the Trunk version, and >> sometimes >> >> when I ask about >> >> > something (like media timeout for SIP trunk), >> then >> >> they say ur asterisk >> >> > vesion should be trunk version. >> >> > >> >> > What is the difference between Trunk version >> and not >> >> Trunk version? And how >> >> > can I obtain the Trunk version? >> >> > >> >> > Regards >> >> > Bilal >> >> > >> >> > >> >> > >> >> > >> >> > >> ___ >> >> > -- Bandwidth and Colocation Provided by >> >> http://www.api-digital.com -- >> >> > >> >> > AstriCon 2008 - September 22 - 25 Phoenix, >> Arizona >> >> > Register Now: http://www.astricon.net >> >> > >> >> > asterisk-users mailing list >> >> > To UNSUBSCRIBE or update options visit: >> >> > >> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > >> > >> > >> > >> > >> > ___ >> > -- Bandwidth and Colocation Provided by >> http://www.api-digital.com -- >> > >> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> > Register Now: http://www.astricon.net >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> >> >> -- >> Erik Anderson >> http://andersonfam.org > > > > -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Trunk and normal
Bilal - I think you're perhaps confusing two meanings of the word "trunk". In this case, "trunk" is referring to the trunk of the SVN development repository, not SIP or IAX trunks. This can be seen as the main development area for asterisk. On Tue, Sep 2, 2008 at 10:22 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote: > Sorry, but I did not find in the below link anything answering the difference > between the trunk and not trunk version? When to use asterisk trunk and > asterisk normal? > > Regards > Bilal > > > --- On Tue, 9/2/08, Dan Julius <[EMAIL PROTECTED]> wrote: > >> From: Dan Julius <[EMAIL PROTECTED]> >> Subject: Re: [asterisk-users] Asterisk Trunk and normal >> To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial >> Discussion" >> Date: Tuesday, September 2, 2008, 9:33 AM >> Hi, >> >> checkout >> http://svnbook.red-bean.com/en/1.4/svn.tour.importing.html#svn.tour.importing.layout >> this explains about versioning >> >> Dan >> >> On Tue, Sep 2, 2008 at 3:56 PM, bilal ghayyad >> <[EMAIL PROTECTED]> wrote: >> >> > Hi List; >> > >> > I see and hear about the Trunk version, and sometimes >> when I ask about >> > something (like media timeout for SIP trunk), then >> they say ur asterisk >> > vesion should be trunk version. >> > >> > What is the difference between Trunk version and not >> Trunk version? And how >> > can I obtain the Trunk version? >> > >> > Regards >> > Bilal >> > >> > >> > >> > >> > ___ >> > -- Bandwidth and Colocation Provided by >> http://www.api-digital.com -- >> > >> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> > Register Now: http://www.astricon.net >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to monitor Asterisk logs ?
On Tue, Jul 15, 2008 at 3:22 PM, Olivier <[EMAIL PROTECTED]> wrote: > Hi, > > How can I be notified anytime a given warning message appears in Asterisk > logs ? Oliver - This is a project I've had my eye on for a while: http://www.splunk.com I've never used it, nor have I set it up, but from reading the feature list, it looks like it's able to keep an eye on any number of log files and notify you if it sees an error. Unless they have built-in asterisk support (which I doubt), I'd bet you'd need to specify some regex rules for what constitutes an "error". Anyway - report back if you end up giving it a try. I've wanted to get it set up for several months now, but haven't been able to due to lack of "play" time in my work schedule. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] First-time queue app: verifying human member?
Good evening all - for the first time, I'm implementing my first-ever queue in asterisk. Overall, it's a pretty simple setup, 4 static members, very low call volume, etc. The one thing that has stumped me so far, though, is the following... This is a queue I'm setting up for contacting our IT support staff off-hours. As such, I've just added the cell phone numbers of our staff as members. I'd like to somehow verify that it's an actual human answering the phone when a member is dialed and not their mobile phone's voicemail. Is that possible? I'd envision just requesting that the member press "1" or something to accept the call. I currently have the timeout in queues.conf set low enough so that the call will never automatically roll over to that member's mobile voicemail, but I can't guaranty that the staff member won't just hit "Ignore" on their phone and send it directly to voicemail. Ideas? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson <[EMAIL PROTECTED]> wrote: > > So now the PBX is over 1.2 Gig for the installation. Typical PBX > installs are under 600 Meg. This makes me wonder about server > stability, reliability and performance as uptime creeps on and user > count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. > Can anyone give me feedback on real world experience with this type of > setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. > Is it better for production to run Openfire on a separate server than the PBX? What's your definition of "better". Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? > My biggest concern is deploying a 100+ user environment with high call > volume and high chat volume. Java seems to be a bit resource hungry > with the user notifications and call pop ups. I would hate to have > the IM server walking over Asterisk and affecting call quality or PBX > stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just "nice" the openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based VoIP client? None of them are very full featured
On Wed, Jun 4, 2008 at 5:52 PM, Bob G <[EMAIL PROTECTED]> wrote: > None of them have features like hold, transfer, voice mail, dtmf, conference > as far as I know none of them has caller ID > > Only 1ezphone.com has all that and the buttons are programmable for CRM > features. Hrm: - no apparent compatibility with any service other than that which is offered via 1ezphone - Frequent spammy emails. - Dubious claims on website: "...we are going to make the only phone portal you will every want." - Some poor person's info revealed on the "User Account" page: http://1ezphone.com/profile.html - Revelation of someone's call history: http://1ezphone.com/callhistory.html# I, for one, won't be giving this a try any time soon. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski <[EMAIL PROTECTED]> wrote: > > Is 384kB up too slow? Probably not. > Is there any guidance for the minimum upload speed for an Asterisk box? I'm guessing this is for just a few calls at a time, correct? I'd guess that rather than these quality issues being caused by cramped bandwidth, they're actually being caused by latency issues. Have you ever checked the latency of the connection between your asterisk server and your SIP/IAX endpoint? If it's really high (say 300ms+) or if the latency is really erratic, you'll have quality issues. You didn't mention whether you are doing traffic shaping on your upstream connection, so I'll assume you're not. That would be something good to look into - with traffic shaping, you can prioritize your VoIP traffic over all other types of network traffic. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Where is SoundPointIPWelcome.wav used?
On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney) <[EMAIL PROTECTED]> wrote: > When I downloaded the sip and bootrom from Polycom website, I noticed a > file called SoundPointIPWelcome.wav. However, I have no idea where and > when it was used. I played the wav file but I have never heard the > phone using this wav file before. Does anyone know what it is used for? It's played at the completion of the boot process. It's always been very quiet on the models I've worked with. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk with lumenvox
On Wed, Apr 2, 2008 at 10:24 PM, Al Baker <[EMAIL PROTECTED]> wrote: > Clearly all of this not feasible in a IVR environment, so, in the > absence of all this, just how good , and how sophisticated of a voice > recognition can one achieve ? Have you ever called Google 411? 1-800-GOOG-411 It'll blow your mind ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with DELL 1600
On Wed, Apr 2, 2008 at 9:37 AM, Ruben Zamora <[EMAIL PROTECTED]> wrote: > > I just want to know if anyone have problems with server DELL 1600, > Like: Hangup Call. Give us some more details of your setup and you'll probably have better chances of getting an answer. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FYI about my Mona Vie business venture
On Mon, Mar 24, 2008 at 1:56 PM, BerkHolz, Steven <[EMAIL PROTECTED]> wrote: > > I am not going to go into a sales pitch. > This is just an FYI to this opportunity. Sorry, but one man's "opportunity" is another man's "sales pitch". > To sign up to be a distributor , which is required to make money, is $54 > A case of Mona Vie is $120. > A case will last 2 people a month. (you only take 2 ounces a day) > > This may seem like a lot, but: > 1. You will not need to buy any vitamins. > 2. My brother-in-law is already making $200 a month, after being in the > system for a month, So his cost for the Mona Vie is covered and he is making > $80 a month. > 3. As more people sign up, the amount he gets back will increase. > > I am very excited with this, both in the health benefits I am already > seeing, and the income potential. Sure looks like a sales pitch to me... This is spam, pure and simple. Please stop abusing the list for your own business "opportunities". -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Wed, Mar 19, 2008 at 4:38 PM, Bill Andersen <[EMAIL PROTECTED]> wrote: > > Although this is a "users" list, I think it is more of a list > for Asterisk "resellers". I'd be interested in how many of you > are simply using Asterisk as your phone system and NOT selling > your services or an Asterisk based solution? > > Anyone? Just a user? /me raises hand. > That being said. As "just" a user of Asterisk, it is clear that > if I want to continue with Asterisk, it looks like I really need > to "learn" the ins-and-outs of Asterisk and ditch my pre-packaged > solution. Off to Amazon for to find TFOT (I want the hard copy :) Agreed - I'm sure you'll be much more happy with the stability of your vanilla asterisk implementation (assuming you're running on a stable OS and server-class hardware) as well as being much more comfortable with what's going on behind the scenes. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RES: phpagi
On Wed, Mar 19, 2008 at 1:31 PM, Carlos Carvalhar <[EMAIL PROTECTED]> wrote: > > But when I download the gz file it doesn't uncompress as php files, the > phpagi-2.14.gz file returns a phpagi-2.14 file...and I tried with winrar and > 7-zip that usually uncompress gzip files without problem. > > How can I get the php files of the class phpagi? > How did you download it? $ wget http://superb-east.dl.sourceforge.net/sourceforge/phpagi/phpagi-2.14.tgz $ tar zxvf phpagi-2.14.tgz $ cd phpagi-2.14 $ ls -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phpagi
On Wed, Mar 19, 2008 at 12:48 PM, Carlos Carvalhar <[EMAIL PROTECTED]> wrote: > > How do I install phpagi? > > http://phpagi.sourceforge.net/ Since phpagi is really just a set of php libraries, all you need to do to install is dump it somewhere and add that location to your php include_path. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid Open Source Edition
On Mon, Mar 17, 2008 at 12:09 PM, Brett Crapser <[EMAIL PROTECTED]> wrote: > > Then I noticed how all the asterisk files/directorys had been 777'ed. Ouch - I think I'll pass as well. -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail Server
On Thu, Mar 13, 2008 at 4:04 PM, Mike Hammett <[EMAIL PROTECTED]> wrote: > > I need to setup a small mail server on a local network. It only needs SMTP > ability as it's just so Asterisk can send out emails. The machine has > sendmail installed. My primary mail server seems to be rejecting the > messages. Some research says something isn't configured properly. What do > I have to do so the outside world accepts emails from my Asterisk box? It > is behind a NAT. Does your ISP provide an SMTP server you can use? If so, it's usually easiest to set that up as a "smarthost" and tell sendmail to send through that server. If this isn't an option, you need to make sure that your asterisk server has a valid publicly-available DNS record (and reverse DNS). That's most likely the reason the remote server is rejecting these emails. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie on VoIP
On Mon, Mar 3, 2008 at 12:40 AM, NOC Ph <[EMAIL PROTECTED]> wrote: > > [NOCPH] I have to open the SIP port and web. Another question, the SIP port > is 5060 UDP, how about the conference? Does it use the same port also? That's a good start, but you'll also need to open the RTP ports as well - these usually fall in the 10k-20k udp range. 5060/udp is used for call signalling only, the actual voice data can use a variety of ports, depending on how you're set up. You can specify what RTP ports you want to use in your rtp.conf. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
On Sun, Mar 2, 2008 at 3:21 AM, Mike <[EMAIL PROTECTED]> wrote: > > Just curious if anyone has suggestions on how one can get a near > FREE(I hope) DID number. Hey Mike - give IPKall a try: http://www.ipkall.com/ They'll give you a free Washington state DID along with free SIP to your asterisk server. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
On Tue, Feb 26, 2008 at 2:10 PM, Matt <[EMAIL PROTECTED]> wrote: > I've had it with Dell server garbage.They seem to change RAID > controllers as much as I change socks, and then the controllers don't work > with Linux, unless you load a new driver.They sell servers with a PCI-e > slot in them, but then you get it and find out the RAID controller is using > the PCI-e slot! Their sales folks are dumber than rocks, and they change > them more often than I change underwear. > [end rant]. Ouch! :-) I can't speak to the PCIe issue, but I've never in my life had compatibility issues with the Dell RAID controllers. What kernel are you on? > Can anyone recommend an IBM or Gateway server that you have used with > Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has > room for one or two PCI-express interface cards? Gateway server? Ew. Have you looked into the new Sun servers? I've been researching them lately, and they have some compelling offerrings. They also offer full support for linux as well... -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I call cheap to UK cell phones
On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: > Greetings, > > How can I call cheap to UK cell phones. I am located in Toronto, Canada, but > need to call UK cell phones both from Toronto and London. I'd guess you could get an account with one of these providers: http://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test
On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki <[EMAIL PROTECTED]> wrote: > checking wheather my mail goes to asterisk users mailling list or not ACK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk Servers. One Conference
On Wed, Feb 20, 2008 at 8:49 PM, Klaverstyn, David C <[EMAIL PROTECTED]> wrote: > > > I currently have about 10 Asterisk servers scattered around the place each > hosting their own dynamic conference centre. Is there any way that when > people join these conference centres on each server that somehow Asterisk > bridges the conference centres on each server to form one large conference? In theory, this wouldn't be difficult at all. I'd imagine it could go something like this: set up one central conference server. Each branch server would call an extension (zap/sip/iax/whatever) on the main server, which in turn would dump it into a certain meetme room. Alternatively, you could have the central server call out to the branch servers and join them to the meetme room. In practice, though, I have no idea how the audio quality would be. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > It also consumes more CPU. True, a fraction more. If you have that little overhead on your server, though, that this would cause a problem, you probably should upgrade your hardware, IMHO. -eriik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
On Thu, Feb 14, 2008 at 8:38 PM, Al lists <[EMAIL PROTECTED]> wrote: > Always rely on free -m to see how much free memory you have not top. You could install and use "htop" - it's a much more functional (and informative) version of top. It shows the difference between shared/buffer/cache memory. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Semi-OT: bluetooth conference phone?
All - I've been trying to pick out a bluetooth conference phone that I could use with a softphone along with my asterisk server. I've been looking at the TrendNet TVP-SP4BK. Have any of you used this device or any other bluetooth conference phone? How have your experiences been? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hookup to cell phone for outbound calls?
On Feb 5, 2008 2:37 PM, Drew Gibson <[EMAIL PROTECTED]> wrote: > > How about http://www.mgamble.ca/oss/iphone_asterisk/ ? Hah! Cool, but quite ridiculous. :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot hear voice through SIP Phone from one side
On Feb 5, 2008 2:32 PM, Sanjoy Rath <[EMAIL PROTECTED]> wrote: > > The Asterisk server is a linux server. There is no firewall between the > servers. It is in a DMZ. My bet is that it's not a *true* DMZ. You're still dealing with NAT, and that's what's causing the one-way audio. This topic has been discussed ad nauseam on the list and is documented quite well on the wiki - search there and you'll most likely find the answers you're looking for. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Compatibility List for Asterisk
It is my understanding that the cast majority of the compatibility issues went away with the recent chipset change on the digium cards. Soa compatibility list really isn't needed. I've run the digium cards on all manner of Dell hardware (from old-school desktops all the way to the high end servers) and have never had issues. On 1/31/08, broadband Voice <[EMAIL PROTECTED]> wrote: > Digium has a compatibility list of servers, however, it has not been updated > since 2006. One of the servers on the list has since been taken out of > production by Dell. Here are the remaining servers on the list: HP Proliant > DL360IBM x206IBM x346 > > > Does anyone has a most recent list and I will be adding the digium cards for > T1 the 220 series with echo cancellation? > -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Join me on Last.fm!
Classy. On Jan 25, 2008 2:37 PM, Sina Owolabi <[EMAIL PROTECTED]> wrote: > > > > Hi asterisk-users@lists.digium.com, > > Add me as a friend on Last.fm so we can share our music taste :) > Check out what I'm listening to. > > > > A personal note from me: > "boo!" > > > > Signing up is free and takes less than a minute. > Just click here to automatically accept my add. > > > > Visit my music profile and leave me a shout! I'll see you around, > - Sina Owolabi > > > > > PS: I'm shina01 on Last.fm. > > > > > You received this message because someone (Sina Owolabi) who knows you sent > you an invitation to join them on Last.fm. Your address was not saved and we > will never contact you unsolicited. For more information, see our privacy > policy at: http://www.last.fm/help/privacy.php. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacement for Allison
On Jan 24, 2008 10:14 AM, Matt <[EMAIL PROTECTED]> wrote: > That worked... hrmm not that great... anyone know of any decent sounding > recording of Allison for Asterisk? What's your definition of "decent sounding"? IMHO and that of many of my co-workers, the default Allison recordings sound great...not sure exactly what you're looking for. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I am having a problem connecting my X-Lite tomy Asterix box
On Jan 20, 2008 8:06 PM, Andrew Ladanowski <[EMAIL PROTECTED]> wrote: > Windows XP. Andrew - you're going to need to get us your sip.conf before we can really assist you any further. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I am having a problem connecting my X-Lite tomy Asterix box
On Jan 20, 2008 7:47 PM, Andrew Ladanowski <[EMAIL PROTECTED]> wrote: > Here are my log information. > [Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from > '"Andrew"' failed for '192.168.3.116' - Device does > not match ACL > [Jan 20 12:35:33] NOTICE[2637] chan_sip.c: Registration from > '"Andrew"' failed for '192.168.3.116' - Device does > not match ACL > > I am not a Linux guy I am a Windows Programmer I can not get to the sip.conf? Are you using asterisk or trixbox? If asterisk, just open up /etc/sip.conf in an editor... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I am having a problem connecting my X-Lite to my Asterix box
On Jan 20, 2008 7:14 PM, Andrew Ladanowski <[EMAIL PROTECTED]> wrote: > > I have added two extentsions. I am try to test connecting X-lite to the > server. > > I have two extension one 1000 with password 1234 and one 2000 with password > 2000. Andrew - could you send us the relevent sections of your sip.conf? That would be quite helpful in helping you troubleshoot this problem. Also, please post any messages that appear on the asterisk console when you try and register your x-lite phone. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production
On Jan 16, 2008 7:28 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > > You can add the raid option for $199. I think I might pickup about ten of > them at this price. I can always resell them as general purpose servers or > even workstations if Asterisk/Zaptel/Linux does not like the boxen. Ahh - nice. That wasn't an option when I ordered the SC440. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production
On Jan 16, 2008 6:39 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > Unbeatable price for a low end Asterisk server (or any server for that > matter) > > http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l=en&oc=bednv4k&s=bsd > > I wonder if anyone has any experience with this box and Digium or Sangoma > hardware? Any compatibility issues? If not, I might stock up on them. Wow - that *is* a great price. I don't have any of this particular box in production, but I do have 2 PowerEdge SC440s (one step up from the T105) running asterisk along with Sangoma PRI cards. They're working great. I really only have two issues with these low-end servers: 1. You can't order 'em with RAID support. I'm getting around this by using software RAID1 in linux, but I'd much prefer having a hardware RAID controller. 2. The Dell DRAC remote management cards aren't compatible with these low-end server motherboards. I've become *completely* addicted to the DRAC cards on the high-end PowerEdges, to the point that I now refuse to order a server without a DRAC card. That said, I'm sure this server would run a small/medium asterisk install just fine. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 10, 2008 8:24 AM, Drew Gibson <[EMAIL PROTECTED]> wrote: > > It has 5 ports! Although the ports are labeled as 1 Internet port and 4 LAN > ports, each can be assigned to a VLAN of your choosing and you can use them > as you please (at least you can under openWRT). Yup - you can do the same with DD-WRT. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 9, 2008 9:40 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > > Heh yeah that's what I was thinking of doing. What's the traffic > shaping like? Can I specify max bandwidth etc or use hfsc shaping? DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB. Here's the dd-wrt wiki page on its QoS implementation: http://www.dd-wrt.com/wiki/index.php/Quality_of_Service Looks like they don't recommend HFSC currently due to some lag issues. That might have been fixed, though, in the more recent firmware builds. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 9, 2008 8:33 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi, > > Does anyone know of a cheap (very cheap) dual port traffic shaping box > (i.e. sub $100) that can be configured for IAX/SIP? Pick up a Linksys WRT54GL and install dd-wrt on it. That will traffic shape any type of traffic you want. I have installed several of these around the country and they work great for prioritizing VoIP traffic. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Question
On Nov 28, 2007 10:52 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote: > > Do sangoma cards use the standard Zaptel drivers? Or do they have to be > compiled externally like Rhino cards? Sangoma maintains a patchset that gets applied to the stock zaptel drivers before compilation. They provide automated tools that will take care of the patching/compiling/installing/configuring for you. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Best firmware for Linksys Router thatis "SIP AWARE"
On Nov 28, 2007 9:44 AM, Dovid B <[EMAIL PROTECTED]> wrote: > > > > So do I. I set SIP to high how ever the calls are still bad. I guess I need > to read up a bit more on the firmware and how to set it up correctly. Are the calls poor quality in both directions or on just one of the "legs" of the call? Implementing QoS on your router will really only help network traffic going *out* of your network. In otherwords, you can really only affect your upload traffic. One thing to consider is that you may just have a poor-quality internet connection. Have you done a VoIP speed test? Here's the one that I use: http://www.voipreview.org/voipspeedtester.aspx This sort of test is ideal for VoIP because unlike most other speed tests, it measures latency, jitter, etc. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
You should be able to issue a "stop gracefully" command to asterisk. That'll cause it to stop accepting new calls, but will let existing calls continue until complete. -erik On Nov 27, 2007 12:06 PM, Alex Balashov <[EMAIL PROTECTED]> wrote: > > In other words, what I need is a way for the upstream switch to somehow > think that the B channels are out of service, but without actually taking > the B channels out of service and dropping the existing calls. > > From within asterisk, zaptel, wanpipe, whatever. Is that possible? > > > On Tue, 27 Nov 2007, Alex Balashov wrote: > > > > > Our provider gives us four PRIs as a trunk group hunt group. Meaning, the > > provider's switch will cycle through B channels in span 1, 2, 3, ... until > > it finds one that is available. > > > > I have moved spans 2-4 onto another machine. But we have one remaining > > box with a PRI full of calls and I don't know what to do with them; the > > box is failing, but dropping them by simply yanking the PRI is not > > acceptable from a business POV. > > > > Sending Congestion() or Busy() in the dial plan wouldn't work because > > the far-end switch would simply pass that onto the subscriber, rather > > interpreting it to mean that the B channel is unavailable and it should > > go on to other T1s in the trunk group. > > > > Any ideas? > > > > > > -- > > Alex Balashov > > Evariste Systems > > Web: http://www.evaristesys.com/ > > Tel: +1-678-954-0670 > > Direct : +1-678-954-0671 > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Alex Balashov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: +1-678-954-0670 > Direct : +1-678-954-0671 > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Filesharing + video + voice supported Soft phone
On Nov 26, 2007 3:07 PM, Bob Gibson <[EMAIL PROTECTED]> wrote: > VMukti.com I have a few comments for you: 1. Your webserver has been throwing 500 errors all afternoon. 2. It appears that all you've been doing with your time all day is spamming the list with "VMukti.com". 3. Do you really think you're convincing any people to check out this product by doing this? Please go away until you can figure out a way to contribute in a meaningful way. -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Best firmware for Linksys Router that is "SIP AWARE"
On Nov 26, 2007 8:29 AM, David Boyd <[EMAIL PROTECTED]> wrote: > > I struggle with the traffic shaping rules, would you be willing to > provide additional details as to what you have done in past? > > Any additional information would be greatly appreciated. Sure - I use the default HTB traffic scheduler. The number one tricky thing about traffic shaping that most people miss is that they don't set their uplink speed correctly. For 99% of the use cases out there, you have no control of your downlink speeds, so there's not a whole lot you can do for that - you really only have control of your uplink packets. So - do a bunch of speed tests and then set your uplink speed to about 80% of your max upload speed. That will ensure that there's always a bit of overhead and that your link itself will never be the uploade bottleneck. After doing this, just start classifying traffic. Here's a synopsis of the rules I use: - DNS - high priority - SIP - express priority - RTP - express priority - HTTP/https - bulk priority - (other p2p applications) - bulk priority Putting those rules in place should make a big difference. You can also specify a specific ethernet jack on the router that will get high priority if that would help in your setup. HTH- Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Best firmware for Linksys Router that is "SIP AWARE"
On Nov 26, 2007 7:51 AM, Dovid B <[EMAIL PROTECTED]> wrote: > Hi, > I am currently playing with DD-WRT and I like it. I am looking for something > that is more "SIP Aware". Anyone know one those that are out there ? Dovid - what exactly are you hoping this "sip aware" firmware will do that dd-wrt doesn't? I've been using dd-wrt in combination with various SIP ITSPs for several years and have had no problems - just add the necessary port forwards and a few traffic shaping rules and it works just fine. I do know that they (the dd-wrt people) have a voip edition of dd-wrt available. I'm not sure what additional functionality it has over the standard version, though. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] p2p t1 with sangoma hw
On Nov 17, 2007 11:49 PM, Michael J. Liberatore <[EMAIL PROTECTED]> wrote: > > I figured that one side would be pri net and the other would be pri cpe, > well I chose pri cpe and the next question was asking for a switch type, > national isdn 2, at&t, nortel, etc - that sounds really wrong. Pick "national" and make sure it's set at both ends. (this is also known as national isdn 2) > So basically I am at a stand still, any help would be great, would it be > pri net on both sides? If its suppsoed to be pri cpe on one side and > pri net on the otherside then what would the switch type be? All > verizon told me is that its b8zs/esf, that's it. One end of your T1 link will need to be pri_net and one will need to be pri_cpe. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - best policy for logs
On Nov 15, 2007 12:55 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote: > > In my experience, it's easier to combine them all into one syslog > server, and then utilize tools to filter them apart when necessary, > since there are more tools to do that than to *combine* them when that > is necessary, which it often is. Agreed - I have all of my servers send their syslogs to /var/log/messages on one central logging server. If you want to examine a device-specific log, just use tail + grep. That said, any system logger worth it's salt will make it extremely easy to have device-specific log files if you prefer. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-stat problem
On Nov 14, 2007 4:15 PM, Richard Cahilig <[EMAIL PROTECTED]> wrote: > Hi, > > I installed asterisk-addons and asterisk-stats, Its working now except > of one problem. The problem is there is no call logs when you open the > cdr report. The message is when you open the cdr report is: - Call > Logs - Back to Top > No data found !!! > 1 / 1 > Did I missed something in the configuration of mysql-addons or > asterisk-stat? Here is my asterisk-stats page: > http://203.115.187.91/cdr, the username is admin and the password is > password. Thank you very much. Richard - just click "search" when you go to one of the report pages. It doesn't do the query manually. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
On Nov 13, 2007 11:44 PM, Mohammad Shokuie <[EMAIL PROTECTED]> wrote: > > HI Erik, > > thanks for your post, Actually im sending new posts not replying but if you > see them correct, how come its wrongly viewed for me. Are you using a > speciall software to view mailing lists? Im just using firefox not a special > one! You're using firefox? How so? I'd recommend either a good email client (Thunderbird) or a good web email interface (gmail). (I'm using gmail's web interface) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
On Nov 13, 2007 11:21 PM, Mohammad Shokuie <[EMAIL PROTECTED]> wrote: > > Anyone knows what is wrong with this mailing list its a while all my new > posts appear as a reply (branch) for others post, is there any hints > i > could prevent this issue?? I believe your posts are all showing up correctly for me. That said, this sort of thing can happen frequently if, instead of composing a new email to the list, you hit "Reply" to an existing message and just change the subject line. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two PRI setup questions
Yep - as Doug mentioned, give esf framing and national switchtype a try. I have a PRI from AT&T in one of my offices, and use this setup. -erik On 11/1/07, Lutgring, Sam <[EMAIL PROTECTED]> wrote: > > > > I am in the process of implementing a new ISDN pri and have a couple of > questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 > interface. The interface looks good and is not showing any errors. Any > help that you can provide would be greatly appreciated. > > 1) What switchtype should be configured in the zapata.conf file when AT&T > is using CUSTOM? My understanding is that this equates to the dms100 in > Asterisk, is this right? The D channel is coming up just fine, but AT&T > tells me that they cannot see the B channels. When I try to make a call I > get a slow busy and the debug shows an ISDN cause code of 34, no circuit > available. > > 2) Is there a way to see the idle status of a B channel? When AT&T tells > me they don't see the B channels coming up, is there a way that I can see > this in Asterisk??? > > Thanks in advance. > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need T1 crossover cable?
On 10/26/07, Michelle Dupuis <[EMAIL PROTECTED]> wrote: > > > I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My > Sangoma A102D shipped with 2 T1 cables - which I assume are straight > through. Do I need to make crossover cables for this scenario? Yes - a crossover *is* needed in this configuration. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
On 10/23/07, Joseph Begumisa <[EMAIL PROTECTED]> wrote: > > Has anyone had any compatibility issues with a TE110P card installed on a > Dell Poweredge 1950? I noted the following error on the LCD display of the > Dell Poweredge 1950: > > E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. > > The Dell hardware owners manual states that it means the system BIOS has > reported a PCI parity error on a component that resides in PCI configuration > space at bus 0, device 4, function 0 and advises that the PCI expansion card > be removed and reseated. I had this error on a 1950 while testing a Sangoma quad-port card. Re-seating the PCI expansion board seemed to solve the problem. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extensions.conf for basic IVR?
On 10/22/07, Vincent <[EMAIL PROTECTED]> wrote: > > 2008 might be a good year to update "* - The future of telephony" :-) Version 2 of TFOT was just released a few weeks ago... http://downloads.oreilly.com/books/9780596510480.pdf -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
On 10/20/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> If you are trying to use non-complied ("XML") profiles... don't even
> bother wasting your time.
Why is that? I'm using the xml-style config and they're working just fine.
--
Erik Anderson
http://andersonfam.org
___
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Re: [asterisk-users] Best USB Handset and Softphone Combination
On 10/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > > Any advice on softphones, handsets, or practical experience with this > sort of deployment? It would be very nice if there was a central way of > provisioning the phones. I've deployed several setups internally using X-Lite and these headsets: http://www.newegg.com/Product/Product.aspx?Item=N82E16826275009 Haven't heard of a single problem thus far. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best USB Handset and Softphone Combination
On 10/19/07, Mike Clark <[EMAIL PROTECTED]> wrote: > > Do they play well with Vista? Hah - I have no idea. We installed Vista on one laptop here when Dell started shipping it. That lasted about 3 days and 10 support tickets from the user. Then we reverted back to XP. Haven't touched Vista since. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
On 10/17/07, shadowym <[EMAIL PROTECTED]> wrote: > Ok so you use templates. I understand that. The problem is some people on > here seem to be claiming they type it all in from scratch in like 3 minutes. Just call me out if you feel the need to. Please don't try and hide behind the "some people on here" type of comments. Call me out directly if you feel the need. I can take it :-) So...I don't feel the need to prove myself to you. I have a fairly good grasp on the conf file syntax, and with a well-thought out and well documented goal, it's not unreasonable for me to say that I can type out a config from scratch in 30 minutes. After working in vim for as long as I have, you learn to use the many shortcuts that it provides for text manipulation, copy buffers, moving blocks of code around, etc. I also use a syntax highlighting rule file for asterisk configs, so any typos I make are immediately evident. It's really remarkable how this discussion has turned into a pissing match. I could really care less if you have a hard time believing my statements. I'm not trying to push CLI on you or anyone. Yes - I recommend that people give it a try before going to a GUI, but I fully recognize that vanilla asterisk text configuration isn't for everyone. -Erik P.S. By the way - don't misquote me. I said nothing about laying down a config in 3 minutes. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
On 10/16/07, shadowym <[EMAIL PROTECTED]> wrote: > So how long would it take you to "vi" a 20 extension office with custom > dialplan involving a medium level of complexity? Including time to debug > etc. Well - there's a large amount of subjectivity in your question, but perhaps I'll answer with "not long". I don't know - 20 sip extensions, maybe 5 minutes. Probably another 30 for the dialplan and debugging. My point still stands - use what you're comfortable with. I spend the vast amount of my day working through an SSH console into various linux servers, so it would only make sense that for me (and many other CLI geeks), it doesn't make sense to use a GUI. I actually get a little put out when I have to switch over to my browser or another GUI tool to get things done. So - the CLI is what works for me. I'm not going to push that on you or anyone as the definitive best management tool for asterisk. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
On 10/16/07, shadowym <[EMAIL PROTECTED]> wrote:
> I don't do text editing so please indulge me. Why would someone want to do
> that when a GUI makes life so much easier?
>
> On a practical note, If someone was deploying 2 or 3 of these a week, most
> of which have 5-10+ extensions doing all kinds of fancy things like call
> queues, parking, forwarding, followme, voicemail to email etc. etc. how
> practical is it to type all this in by hand making sure to get ever single
> space, ".", ",", "{}", "[]" etc. exactly right which NEVER happens. So then
> you have to spend more time debugging the conf files.
>
> Even with a bunch of pre-made templates it seems like an awful lot of
> unnecessary heavy lifting when a GUI can make it so much easier and
> efficient.
This is *very* much a "to each their own" issue. You say that a web
GUI is more efficient - I say that vi is more effecient. You say that
using a text editor is more error-prone - I say that a web GUI is more
likely to mess things up in a difficult way to troubleshoot.
Use what works for you and don't worry about it.
:-)
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Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On 10/12/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > > I wouldn't be too happy about a system with a > loadavg of 3. The system he mentioned had 8 cores, though. So a load average of 3 is less than 50% usage. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
On 10/12/07, D4rk F1ber <[EMAIL PROTECTED]> wrote: > > Curious what others are using, and if anyone can make some > recommendations? Not sure if this has been covered already on the > list, and not sure if recommending companies are allowed, so maybe I > need get replies off list? There are quite literally hundreds of VoIP service providers out there: Here's a list of some of them: http://voip-info.org/wiki/view/VOIP+Service+Providers+Residential Billing schemes usually fall into one of two categories. They'll either bill you a flat monthly fee for an "unlimited" plan or one with a large number of minutes. Or...they'll bill you on a per-minute, usage-only basis. The only provider I've had direct experience with is Teliax. I'm on an outgoing-only plan with them and it's been perfect so far. They bill something like $0.025/minute. If you want incoming calls as well, there's a per-month DID charge. If you are just wanting to receive incoming calls, check out IPKall - they'll give you a DID and a SIP trunk to your PBX for incoming calls. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On 10/12/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > > I don't think there is a formula like > cpu usage = loadavg / #cpus > > A loadavg of 3 says that there are 3 processes waiting to > be executed. > > Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation "For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage." - Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On 10/11/07, Raúl Gómez C. <[EMAIL PROTECTED]> wrote: > > > At this point I was wondering if Asterisk gets real benefits on systems with > several cores (up to 8 in Dell PE2950) for a system that will handle up to > 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax > (Sangoma A400D PCI card). For this load level (even with high-load transcoding), a multi-core machine certainly would not be needed. That said, it certainly wouldn't hurt anything to add on extra cores, especially if they're free ;-) -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does the future arrive?
On 10/9/07, Hans Witvliet <[EMAIL PROTECTED]> wrote: > Hi all, > > Probably this is the wrong place to ask, > but is there an estimated time of arrival of the future? > i.e. TFOT--next generation dealing with * -1.4 > > I attended a workshop some time ago, and the book was part of the > package The Future, my friend, is here. http://downloads.oreilly.com/books/9780596510480.pdf Enjoy! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone using SIP trunks from Time Warner Telecom?
On 10/8/07, Forrest Beck <[EMAIL PROTECTED]> wrote: > > I was told that Asterisk was supported when we looked at the service. Hey Forrest - thanks for the information. Might you be able to send along the contact information for the TW rep who told you that asterisk was supported? I've been in conversation with our Sales rep today, and he's quite adamant that they currently only support Cisco Call Manager and CCM Express. I believe they're using CCM to provice the SIP trunks - if this is indeed the case, I don't see interoperability with asterisk as a problem. Thanks -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone using SIP trunks from Time Warner Telecom?
I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it wouldn't be difficult for them to honor the QoS flags we set on the SIP/RTP packets. Anyway - has anyone had any experience with Time Warner's SIP trunks? Officially it seems that they only support CCM and Avaya PBXs, but I'd imagine asterisk could be massaged into working just fine. Thoughts? I'm currently negotiating with our TW Sales Rep. to see if they could provision a few test DIDs on a SIP trunk so I can verify compatibility, so I *should* hopefully have answers soon for many of these questions. Thanks! -erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
On 10/6/07, Matt Florell <[EMAIL PROTECTED]> wrote: > > Do not use Dell. I have had issues with both Sangoma and Digium cards > on multiple brand-new Dell servers. This is the only vendor that has > consistently given me problems with telco-interface cards. I'll have to refute this. Every single asterisk system I've put together has been on a Dell hardware. I use Sangoma linecards in all of my systems. I'm running several of the lower-end Dell "SC" server series, as well as several servers running on higher-end PowerEdge 2950, 2650, and 1950 hardware. I haven't had a single problem on any of them, audio quality-wise or stabilty-wise. -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How To Transfer Asterisk Installation to a Different Machine
On 10/1/07, Robert DeVries <[EMAIL PROTECTED]> wrote: > > Anyone have a list of the files that would need to be moved? (Obviously the > *.conf files in the Asterisk directory, I can think of some others, but if > someone ever did a list that would be a great help.) You'll probably want to move the subdirs of /var/spool/asterisk that apply to your install as well. -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
On 9/30/07, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > > I don't know about you, but I've had nothing but very good results with > VOIPSupply. I didnt do huge business with them, but I have purchased new and > refurb polycoms from them without so much as an ounce of pain. Ditto - I've never had a single problem with them. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Digium Hardware?
On 9/28/07, William Stillwell (Ki4swy) <[EMAIL PROTECTED]> wrote: > What is the recommend Digium Card for a PRI in NA ? William - this has been discussed ad nauseam on the list recently. Some will suggest that you forget Digium and use instead a Sangoma card. I personally have only used Sangoma cards, so I can't speak to the quality of any other brands. My feeling, however, is that you'll have an equally pleasant experience regardless of whether you choose Digium or Sangoma. So - to answer your question directly, there really aren't that many Digium cards to choose from: http://www.digium.com/en/products/hardware/digitalcards.php You need to choose how many T1 spans you need and whether you want a hardware EC chip on the card. I'm not sure if Digium sells a PCIe version of their single-port card. HTH- Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
On 9/27/07, Doug <[EMAIL PROTECTED]> wrote: > http://www.atacomm.com/ Heh - yah I pulled up their website earlier today with the hopes of purchasing a Polycom SIP conference phone. Oh well... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
On 9/24/07, Steve Davies <[EMAIL PROTECTED]> wrote: > > The phones can send a parameter to the provisioning server to indicate > that they want an "Update" if they do this, and you send no network or > other major config parameters, the phone does not reboot. > > Look at the Linksys provisioning PDF for more details of the parameter. Really? I've been through this document several times looking for something like this and haven't found a single reference to it. Could you provide more details or at least a page number in the Linksys SPA provisioning doc? Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf best practices?
On 9/18/07, C F <[EMAIL PROTECTED]> wrote: > Use the extension, and use grep to determine which account uses which > phone. For example I provision my spa9xx phones from a subdirectory on > apache called spa which on slackware is at: /var/www/htdocs/spa/ > doing: > grep 123 /var/www/htdocs/spa/* will tell you which phone it is. That's a great idea - probably seems like the most simple option. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf best practices?
All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI debug in Asterisk
On 9/18/07, Arpit Mehta <[EMAIL PROTECTED]> wrote: > Hi all, > > Does Asterisk contain a full fledged ISDN packet sniffer. By giving the > command > " pri intense debug span 1 " , does it debug every packet received > (control and voice/data packets) ? To get the equivalent of a packet sniffer, you'll need to go to a lower-level tool than asterisk. For sangoma cards, you can use the `wanpipemon` command to do a packet dump. I'm not sure what the equivalent for Digium cards is, but I'm sure it's possible. -erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different Networks
On 9/7/07, Mike Hammett <[EMAIL PROTECTED]> wrote: > If it has nothing to do with Asterisk, then why does every other device work > as its supposed to? You never answered as to whether or not you're able to get out past your gateway with any other network applications on your asterisk server. Fire up [links/lynx] and pull up www.google.com. Does it work? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different Networks
On 9/6/07, Mike Hammett <[EMAIL PROTECTED]> wrote: > > I have multiple upstreams in my office. The primary upstream is having some > issues with latency\jitter. I want to move the VoIP traffic to another > interface. > > I have the router set to send all traffic destined for "local" networks out > the respective interfaces. Traffic destined to the Internet goes out one of > the upstreams. > > I can do this on a per-IP basis and have successfully done so in testing on > my laptop and a couple other machines. I also have it in production for an > ATA. > > I also switch all devices to use another upstream with the failure of the > primary ISP. > > Again, this works with everything but the Asterisk server. > > The internal Asterisk server cannot connect to the Asterisk server out on > the public Internet. How do I investigate this? Mike - there's no reason this routing problem would have anything to do with asterisk itself.Have you tried running links (or another text web browser) on the asterisk server to see if you're able to get traffic past the gateway? Do you have the default gateway and/or routing tables configured correctly on the asterisk server? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card (off list)
On 8/28/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > Also, you seemed to miss Brian's main point, keeping calls up is not > going to tax your box or prove anything really, you want to create as > many short calls as possible. Run BOINC in the background for a CPU > burn-in test. Well - with this situation, I'm not so much concerned about CPU load as I am PCI bus stability. When getting this server set up, the Dell BMC caught a few odd PCI errors. I haven't been able to reproduce them. At the time the errors occurred, I didn't have any active spans up, so I couldn't determine if the PCI errors would possibly cause calls to drop. This is my main reasoning for wanting long-running calls - I'd let the calls run until I see this error come up again and then see if any of the dropped. Perhaps my troubleshooting logic is flawed, though... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card (off list)
On 8/28/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > Another more creative tool would be to place an ad in the Penny Saver or > whatever your local equivalent is for a "free 42 inch LCD TV, you haul" > and list your number. I bet that would generate alot of calls. You could > put them through and IVR, then a queue, and finally a meetme room. Hah - I admire your creativity :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card (off list)
On 8/28/07, Brian West <[EMAIL PROTECTED]> wrote: > What exactly are your needs? I can provide you some sipp scripts > that might help you. Brian - thanks for the reply. If you read my email, I believe I make it fairly clear what my needs are. I have a 4-port Sangoma PRI card installed. Crossover cables are connected between ports 1->2 and ports 3->4. I'd like to generate a bunch of calls over those spans. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load testing/burn-in for Sangoma quad PRI card
Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1->2 and 3->4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1->2 span and 23 over the 3->4 span. I'd like to start the calls and then let them stay connected for several days to make sure things are in order. This number of calls would be a *lot* higher load than this system would ever see, but I just want to be safe. Is there currently any script out there that would facilitate this sort of testing? Here's my current config: linux-2.6.21 asterisk-1.4.10 zaptel-1.4.4 wanpipe-3.1.3 libpri-1.4.1 Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk multiport
Off the cuff, I can't recall if asterisk can listen for (in this case I assume) SIP on multiple ports. It would be quite easy to do this redirection with iptables, though. On 8/15/07, Walter Willis <[EMAIL PROTECTED]> wrote: > hot to asterisk multiport...??? > example 5060, 5061, 5080 > -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Modules and Sip Outbound
On 8/13/07, John Meksavan <[EMAIL PROTECTED]> wrote:
> Asterisk Users,
>
> I have never done a dial plan for this scenario before. Is it possible to
> have Sip Phones make outbound calls through the PSTN? What would the call
> routing/dial plan would look like?
Yes - certainly possible. There's nothing different about the call
routing going from SIP->Zap as from SIP->SIP really. Assuming that
you already have your zaptel device(s) configured correctly, something
like this in your dialplan is all you'll need. This also assumes you
want to dial "9" to get an outside line.
[globals]
OUTBOUND-TRUNK=Zap/g0
[outbound]
exten => _9NXXNXX,1,Dial(${OUTBOUND-TRUNK}/${EXTEN:1})
-Erik
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Re: [asterisk-users] How to use OpenVPN with Asterisk
On 8/9/07, MOSBAH ABDELKADER <[EMAIL PROTECTED]> wrote: > Hello, > > Have i to install OpenVPN in each Asterisk server or it is enough to install > it in one side only?. Both. You best take any further questions to the OpenVPN mailing lists. You'll get much better information and help there. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
On 8/9/07, MOSBAH ABDELKADER <[EMAIL PROTECTED]> wrote: > > Is the OpenVPN the ideal solution to set a tunnel between two asterisk > servers or there is a better solution. There is no global "ideal" solution. The solution that is ideal for *you* depends on many factors: - What will the tunnel be used for? - How secure do you want the tunnel to be? - What is your level of familiarity with linux networking/routing? - etc., etc. Pick something - be it vtund, OpenVPN, Racoon, IAX trunk encryption, ssh tunnels, etc. Play with it and see if it's something that will fulfill your requirements. If so, put it into production and be done with it. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Erik Anderson <[EMAIL PROTECTED]> wrote: > I've been going back and forth with my telco for several days, trying > different configurations to get a new PRI to come up. The bchannels > are all up and the T1 is not in alarm status. The dchannel refuses to > come up however. We've tried ni2, qsig, and now dms100 for the > switchtype. The telco tech I've been working with says that he's been > sending "reset all channels" signals to my system, to which he's > getting an "establish remote" response from my asterisk box. I've > been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel > this whole time and have yet to see a single incoming packet. I > believe I *should* be seeing an incoming packet when he sends the > reset, correct? Is there any way to do a completely raw dump of the > d-channel? Thanks to everyone who offered suggestions on how to troubleshoot this issue. After working with the telco for over a week on this, I finally got them to admit today that they have a configuration problem. I had been telling this since day 1, but they didn't listen to me. Their change in perspective came when they had a tech come on-site with a PRI emulator device. He connected that directly to my asterisk server and was able to make calls with no issues whatsoever. Fortunately after this final test, they admitted that the problem must be on their end. Hopefully they'll get it sorted today. As an aside, I had a quick question regarding smartjacks. Is there a jumper or something on the smartjack itself to change from an old-style E&M T1 to a PRI? I'd think that change would happen in the telco's switch, but I just thought it might be a possibility. In my case, as I stated in my original email, the bchannels come up fine, but not the dchannel. This makes me think it could be something simple... -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 or analog line
On 8/7/07, fateme fatah <[EMAIL PROTECTED]> wrote: > Hi: > I want to have conference call(meetme) service with asterisk and 30 > users.Now do I use 1E1 or 30 analog lines with due attention to high price > of E1 line?And which interface card do I use? I'm not sure what analog prices are in your area, but I'd be fairly certain that the costs for 30 analog lines would be *far* more expensive than an E1 line. Cost aside, terminating and managing 30 analog lines will be a big pain. Go with the E1 - it'll save you many many headaches. I've only worked with Sangoma interface cards, but they've worked flawlessly for me so far. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchtype
On 8/7/07, Jeremy Mann <[EMAIL PROTECTED]> wrote: > > In Zapata.conf, if my PRI is NI-2 configured, do I still use > switchtype=national ? Yup: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf#ISDNPRISwitchConfiguration ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/7/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote: > In your wanpipe1.conf see if you have > > TDMV_DCHAN = 0 Nope. I have it set to 24. -erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Anthony Francis <[EMAIL PROTECTED]> wrote: > Yeah you are sending the SABME's because you think you are the master, > they are not replaying with a UA because they think they are the master, > you should def be pri_cpe. Tried it...no go. > There is one other potential cause here, you may not have had the > sangoma install patch and rebuild zaptel. Not doing that can cause a D > channel lockout on your end, but the provider should be able to see the > the D is in lockout. I re-patched zaptel, compiled, and re-installed. No difference. I think I'm just going to have to wait until tomorrow when I can get both Sangoma and the telco on the phone. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Anthony Francis <[EMAIL PROTECTED]> wrote: > You should never be the signaling source, you are always a slave to the > provider, go with pri_cpe and see if things go better. That's what I've experienced in the past, but they were adamant about me being the network end. I tried switching to cpe for the heck of it, but that didn't help... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Anthony Francis <[EMAIL PROTECTED]> wrote: > also in asterisk do: > pri intense debug span 1 > Then you should see UA's and SABME's, If you don't, your not talking to > them. I see plenty of SABMEs, but nothing else: > [ 02 01 7f ] > Unnumbered frame: > SAPI: 00 C/R: 1 EA: 0 > TEI: 000EA: 1 > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode > extended) ] > 0 bytes of data Sending Set Asynchronous Balanced Mode Extended > [ 02 01 7f ] > Unnumbered frame: > SAPI: 00 C/R: 1 EA: 0 > TEI: 000EA: 1 > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode > extended) ] > 0 bytes of data Sending Set Asynchronous Balanced Mode Extended lpdlnx04*CLI> pri > [ 02 01 7f ] lpdlnx04*CLI> pri > Unnumbered frame: > SAPI: 00 C/R: 1 EA: 0 > TEI: 000EA: 1 > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode > extended) ] > 0 bytes of data Sending Set Asynchronous Balanced Mode Extended ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > > I have done a conference call with the telco guy, myself, and a Sangoma > tech at the same time. I was just quite and let them battle it out. It > turned out to be a telco issue but the Global Crossing tech wanted to > blame me and my equipment. He ate a little humble pie on that one. > > If I were you, I would call Sangoma, sometimes the French Canadian > accent is tough but if you give them root, it shouldn't be that bad. > They have several techs and any one of them should be able to help. This sounds like a great idea - I'm going to try and get Sangoma and the telco tech on the horn at the same time tomorrow. -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Darryl Dunkin <[EMAIL PROTECTED]> wrote: > wanpipemon is the way to do it as far as I know. > > For starters, what do your zaptel/zapata configs look like? lpdlnx04*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: Nortel DMS100 Type: Network I know it's odd, but the telco instructed me to set my equipment as the network end...hence pri_net: /etc/zaptel.conf loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:10 bus:2 span: 1] span=1,1,0,esf,b8zs bchan=1-8 dchan=24 /etc/asterisk/zapata.conf [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;;Sangoma A102 port 1 [slot:10 bus:2 span: 1] switchtype=dms100 context=from-pstn group=1 signalling=pri_net channel => 1-8 There you go. As an aside, turns out that it's a national holiday in CA, so the Sangoma support guys are on vacation for the day. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > Call Sangoma and give them root if you can. They will fix it quickly or > at least give you ammunition that it is the telco's issue. Good idea - I just emailed them. Hopefully they'll respond quickly. My normal contact there (Jignesh) is either out of the office today or at least he forgot to start up MSN this morning, as he's showing offline. Hopefully he's not the only tech support guy there. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] low-level dump for PRI dchan debugging
I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels" signals to my system, to which he's getting an "establish remote" response from my asterisk box. I've been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel this whole time and have yet to see a single incoming packet. I believe I *should* be seeing an incoming packet when he sends the reset, correct? Is there any way to do a completely raw dump of the d-channel? Here are my specs: linux-2.6.16 libpri-1.3.5 zaptel-1.2.19 asterisk-1.2.21.1 The PRI interface is a Sangoma A102...it's running the latest firmware and I'm running wanpipe-2.3.4-12 for the sangoma drivers. Any ideas? -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri "call by call" trunking?
On 8/2/07, Don Kelly <[EMAIL PROTECTED]> wrote: > Hi, Erik, > > Never heard of call-by-call trunking. > > Are you in Minnesota? What carrier are you using? Yes I am...this is for one of our branch offices, though, outside of Boston, MA. -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri "call by call" trunking?
On 8/1/07, John covici <[EMAIL PROTECTED]> wrote: > I had some troubles -- try setting the timing parameter to 0 (second > one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for timing. That's not the case in my situation, as I need to be syncing with the telco's clock. That said, in the interest of troubleshooting, I did try setting it to zero - this didn't fix the problem. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri "call by call" trunking?
On 8/1/07, C F <[EMAIL PROTECTED]> wrote: > what channel are they putting the Dchannel on? > Post your zapata.conf and zaptel.conf The D channel is on 24. zaptel.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf lpdlnx04 asterisk # cat zapata.conf ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:10 bus:2 span: 1] switchtype=national context=from-pstn group=1 signalling=pri_cpe channel => 1-23 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pri "call by call" trunking?
I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on "call-by-call" trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. -Erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
