Re: [Asterisk-Users] LiveVoip is Bankrupt
On Monday 27 June 2005 20:04, Robert Webb wrote: I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. In a Google search box: site:lists.digium.com What you are searching for The problem many newbies faces is TOO MUCH information. Not being able to see the trees because of the forest basically. It does not matter either if it has been discussed until someone went crazy or died. The reason it keeps coming up is because it has not been solved. I totally agree with why. I sure don't want to be the one babysitting them. These posts were simply pointing out what I think, as a former educator, is part of the problem. Something which is not that hard to do. And indeed during some spare times I may put together something which is a lower gradient for those who have a hard time getting it. I sure would like to. Now it could very well be that many of these people never get anywhere because it's just too hard for them. But I know when I started a few years back, that a lot of the howto's have a stiff gradient. It skips pieces of information, assumes knowledge which is hard to come by and so on. Standard stuff. I'm not assuming or expecting that anyone is going to act on what I'm saying. If it was easy someone would have already implemented it. But I am saying that I see there are things that CAN be done which will make it easier. And if it makes it easier, this list will have less stupid and repetitive questions. More people will win using Asterisk and we should all win. (Except those who prefer fewer people competed in this arena. And there are a few here who are happy it's hard for others to take part of the fruit. There always are.) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipJet TOS (was Teliax and also LiveVoip)
On Tuesday 28 June 2005 15:02, Wiley Siler wrote: One would assume they have better things to do as they are quite busy. I think this is just a proactive measure meaning they say you cannot do it upfront so that in the event of a problem, it was predeclared. As to the rest of the TOS, I could be wrong but I got the impression that the owner of VoipJet speaks English as a second language due to some email exchanges. If that is the case, the TOS issue can just be one of cultural and language variation. I could be totally wrong though... Regardless, one can assume that the ability to listen to calls is there. And quite frankly is usually illegal. AOL used to get around that by having their EULA say that AOL owns all the emails etc. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
On Monday 27 June 2005 14:31, Michael Di Martino wrote: If this list spent at least half the time on helping other asterisk admins as it does on trivial things like LiveVoips bankruptcy it just might be a great list. As it stands now this list is kind of useless. Most request for assistance with asterisk problems go unresolved of unanswered. Lists with this number of new members have a repetiveness of the same questions which people sometimes get tired of answering. Which is too bad. However, even though it seemingly does not directly aid asterisk users it does so indirectly. People on this list grow into becoming lemonade stand operators and maybe even bigger service providers. It is often done because people realize that Wow, I can do it too!. Unfortunately it's not something that lives in the world of the Internet, but enters the heavily controlled area of phones. An area droght with difficulties for any newcomer. The thread is showing and giving reason to be a bit better prepared when entering into this particular service industry. As such it is of great importance to those wise enough to take note. For the rest it's just noise. One could probably argue effectively for an Asterisk-Basic list. Or an Asterisk-Advanced user list. Something that makes it easier to get started without being overwhelmed by 10,000-15,000 users posts. A place that frequently posted links to the beginner pages on the wiki. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
On Monday 27 June 2005 17:26, Andrew Kohlsmith wrote: On Monday 27 June 2005 15:46, steve szmidt wrote: One could probably argue effectively for an Asterisk-Basic list. Or an Asterisk-Advanced user list. Something that makes it easier to get started without being overwhelmed by 10,000-15,000 users posts. A place that frequently posted links to the beginner pages on the wiki. We've effectively argued it to death many many times over the course of the last few years. Check the archives -- it's been thought up and re-thought up and dismissed each and every time. Yes, I remember. Basic issue: nobody will want to sit on the newbie list because they'll end up answering all the same questions over and over since nobody really seems to want to read for themselves. Exactly. Though I think we could have more success if we had a dumb'd down version of the wiki with very few options. Maybe along the line Minimum-1, A-Bit-More-2 and A-Little-Bit -More-3. The wiki could do with a MySQL web type layout. I'm thinking of the many examples part. But I sure don't have the time to do it. But I would be tempted. It's the same argument that comes around for forums, except that last time I think we actually witnessed a man lose his mind on the mailing list. That was entertaining. :-) Hehe... we don't see that much fun every day. Which is probably why it was entertaining. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
On Sunday 26 June 2005 15:20, Andrew Kohlsmith wrote: On Sunday 26 June 2005 14:32, Orlando Guitián wrote: If anybody is interested, we offer a VoIP solution. we have manufactured our own equipment and network from the ground up. The service has been selling successfully selling for over one year (both domestic and internation). If interested, let know and i will send you pricing and information, [EMAIL PROTECTED] I'm sorry but anyone selling their service for over a year without bothering to mention their company name and indeed, using an msn account already has me sufficiently suspicious to decide against giving them any money. It would seem that people just don't realize how it makes them look. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Configuration Best Practice
On Tuesday 21 June 2005 10:38, Goolsby, Daniel S (Daniel) wrote: Could you just configure the extention to be a ring group instead of an actual extention, or ring queue.. then have his phone/laptop log in whenever he's at the office/coffee shop? As someone else pointed out if you want to keep it simple just use: exten = 1234,2,Dial(SIP/1234SIP/1235,15) It will dial all their extensions. Why make it more complex? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with pf and asterisk
On Tuesday 14 June 2005 02:04, Michiel van Baak wrote: On 20:00, Mon 13 Jun 05, Frank Cases wrote: current setup SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address or #2 asterisk trying to get back to me as 192.168 on public internet.. got canreinvite=yes and no. nat=yes qualify=1000 externaladdr=IP of (em1) localnet=172.16.0.0/12 Try to set the externaladdr to the IP of xl0. That did the trick for me here. Well, you better make it for what your interface is. Type ifconfig -a to see what type of NICs you have. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk code
On Monday 13 June 2005 12:06, Matt Riddell wrote: Race Vanderdecken wrote: Also subscribe to the asterisk-dev mail list. Watch it for a couple of days before you ask a question or they will eat your lunch. Or even more likely, eat you for lunch! :D Phew! I thought lunches was going to start disappearing... -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiPSupply Dot Com
On Sunday 05 June 2005 15:34, C F wrote: On 6/3/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: Let's clear this up once and for all... Yeah it was cleared of before, you fucking idiot liar, politician, go sun bath with your Nazi friend Kennedy and don't bore us with your lies and rantings. Cry baby. Alright! I've not bothered to keep up with this thread but it's surely getting out of line with this kind of language. I'm sure 98% here, including Digium, did not put up the list for this kind of verbal abuse from anyone. It's true we can delete it, but not until after it's too late. So please spare us, be so kind as to take it off list. Both of you. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP or IAX
On Thursday 02 June 2005 02:48, Sandeep A.S wrote: For bridging VOIP with PSTN Lines Which one is giving better performance SIP or IAX ? I am looking at a result without NAT in picture ? Can some body give details from experiance ? Also with single SIP/IAX channel can I use more than one call at a time ? From a security viewpoint SIP is not even a good option. If you have all the calls terminate in the same location you can turn on trunking with IAX2, and so cut your bandwidth considerably. Networks don't have channels, as Olle pointed out. It's a matter of available bandwidth. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server
On Thursday 02 June 2005 03:14, Tony Mountifield wrote: I find DHCP on my LAN extremely useful for both my and visiting laptops. Any machine that will be using my LAN regularly gets a static entry in /etc/dhcpd.conf so it will always get the same IP address. It also gets an entry in my local DNS. That would address Steve's concerns regarding traceability. Nope, different issue. The reason not to have random IP addresses assigned to computers on a LAN, which you manage, is so that you can eliminate a source of confusion. It gives a point of stability when the same computer keeps the same IP. You soon get familiar with who's doing what, and so new or odd behavior is easier to recognize. As far as tracing goes you can always look in the dhcp server log and see who had what IP at what time. Which is a total pain, if you notice something going on and you want to quickly see who is involved. Granted it's much nicer to not have to configure any computers. So it comes down to a balance, which is exactly what security is all about. Balancing risks. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server
On Thursday 02 June 2005 10:17, Dave Cotton wrote:
What's wrong with :-
host W2K {
hardware ethernet 00:30:1B:AC:39:E3;
fixed-address 192.168.1.130;
}
this box always gets the same IP and I know who's got what.
Nothing, that's really how they all should be done, to avoid anyone leaving a
hostile device on your network. (Since I have not used a dhcp server since
the -90's I did not remember/realize you can assign IP by MAC.)
--
Steve Szmidt
They that would give up essential liberty for temporary safety
deserve neither liberty nor safety.
Benjamin Franklin
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Re: [Asterisk-Users] Re: Asterisk Box as a Router, Firewall =?utf-8?q?and=09DHCP?= Server
On Thursday 02 June 2005 11:18, Dave Cotton wrote:
On Thu, 2005-06-02 at 10:09 -0500, Kristian Kielhofner wrote:
Dave Cotton wrote:
Have another look at it because this only scratches the surface. I
love DHCP.
I second this completely. ISC DHCP allows you to do some crazy
things... Start doing diskless clients with PXE/Etherboot and you can
start to realize how truely flexible and powerful it is!
host ws101 {
hardware ethernet 00:50:04:45:39:EC;
fixed-address 192.168.1.101;
if substring(option vendor-class-identifier, 0, 9) =
PXEClient {
filename /3c905c-tpo.lzpxe;
}else if substring (option vendor-class-identifier, 0, 9) =
Etherboot {
filename /lts/vmlinuz.ltsp;
option vendor-encapsulated-options
3c:09:45:74:68:65:72:62:6f:6f:74:ff;
}
}
Exactly!!
Mmm. I hate to admit it but you _might_ have a valid use model here. : )
Suppose I can always come back with yet another service that needs to be
running and open for hacks. But it does look pretty interesting... Thanks!
--
Steve Szmidt
They that would give up essential liberty for temporary safety
deserve neither liberty nor safety.
Benjamin Franklin
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Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server
On Wednesday 01 June 2005 23:27, Samy Antoun wrote: Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. It really depends on what kind of load that cpu is going to have. There's no technical problems with doing the above. Except I don't see the point with having a dhcp server, unless you are an ISP. My anti dhcp speech: DHCP makes it hard to see who's connection/packets you are looking at when you are checking out what is going on on the LAN. You won't be able to learn the typical activities that people do, and so be able to recognise odd behavor. Every time you see an IP you have to figure out who it belongs to. The work to add a specific IP is so short anyway. Router and Firewall services are not very consuming, nor is a DHCP server. But the idea is that if you start having quality problems or you are going to push the box, you'd be smart to have absolutely nothing running but what you actually need. A well configured Linux box is usually better than an off the shelve dedicated appliance. They have too many money vs technical issues and technology suffers. That's pretty much true with all of them. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rack Mount Server Recommendations
On Thursday 19 May 2005 15:55, Bruce Komito wrote: We've tried a lot of different types of boxes, but the best I've found so far has been from SuperMicro. Contact me off list for more specifics. Bruce Komito Why this would be good for the list to see. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF intermittently stops working
On Monday 18 April 2005 22:30, Pedro wrote: Every so often I get a report from a customer that DTMF stops working while checking voicemail. The customer has to hang up and check for messages again. I have actually had this happen to me twice in the past 6 months so I know it does happen, just not very often. So far, the only incidents have been with Cisco 7960's. I was just wondering if anyone had noticed this behavior in their environment. We are using ulaw and rfc2833 with the following configuration (Asterisk CVS-v1-0-11/12/04): With Snom too... -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
On Friday 01 April 2005 02:40, Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. Olle, you better take a break! For the rest of you, good luck! You'll need it. I think finally the Danish Elephant beer that is so strong has gone to Olle's head. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On Thursday 31 March 2005 02:43, Brian Capouch wrote: Isamar Maia wrote: Technically speaking not. But Sangoma's support seems to be pretty much better. My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. They're glad to use Asterisk as a selling point for their hardware, but unwilling to donate anything back to the Asterisk community. It really does become an interesting debate. Do you lower your own ability to survive by using a lower quality product/service, to help ensure that the main product continues. Or do you help the main product survive by putting yourself at risk? For better or worse we are all also the effect of Digium's policies and decisions. Not to say that they have not done an outstanding job getting Asterisk to what it is. Likewise Digium is at the effect of what the community does. In the long run one needs to find a balance where everyone can win. Usually that is done by plain market influence. If they don't buy it, you won't be able to make it for very long. Indeed we all would be poorer if Digium could not continue the work. But so too, do they need to ensure that they are staying close to community needs, while making sure they DO make the right decisions. I think it's fair to say that Digium is more right than wrong, as their course have taken them this far. One does however need to reevaluate positions and directions every now and then, and be willing to change course should it so require. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. My view of Asterisk has made me put my money where my mouth is by betting the farm on Asterisk. I have put everything I have into a position of making a living with Asterisk, so I too depend on it to survive. But in the end I have to ensure that my decisions keep food on our table. Whether I choose Sangoma or Digium cards will be based on what I perceive to be the most long term survival thing to do. Of course, if I end up making a good business out of Sangoma and Asterisk, nothing will stop me from paying license fee's to Digium, which will be more profitable than selling me a card anyway. So I see that Digium should be making enough money from all of us, each contributing in a different way. In fact at this point Asterisk is poised to become a major influence in the market as people world wide is waking up to it's potential. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email forum??
On Thursday 31 March 2005 10:26, Chuck Bunn wrote: Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any forums that are well used or that might even convert this email in a true forum that is searchable and that doesn't require me downloading every email. Before you go and rant on me go see how Mambo Server does it at http://forum.mamboserver.com. The forums are easy to use and thus are easy to participate in. I use mozilla Thunderbird and I have setup filters and all but it still is a pain to use this outdated email forum. Thanks Well I for one would not like to see it change from mailing list. It's much easier to work with and search as I need to. Web lists are too slow to move around for one. I for one would not frequent a forum unless I had no other choice. It sounds like you have a crappy mail client and are trying to work around it. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email
On Thursday 31 March 2005 16:46, Andrew Kohlsmith wrote: On March 31, 2005 01:19 pm, [EMAIL PROTECTED] wrote: Any decent on-line forum would be much better than these digium email lists. The lists are poorly formatted, there is no easy way to post code, you cannot neatly quote anyone, the s earch function in the archive is elementary at best, there is no possibility for active use rs to moderate their area of interest, there is no private messaging, the list goes on and on. Are you on crack? Posting code is simple. Just post it. Quoting? You gotta be kidding, get a BASIC email client, what are you using, telnet? Private messaging? Send the email to the person, not the list. Moderation? Are you a child? Do you need to be moderated? Well Andrew, I got to give it to you! You made me laugh out loud! Too funny! : ) Haha, I had such questions as to what is up with this, and you hit it dead on the nail. Bass, really. What are you using. How can you have these issues? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email forum??
On Thursday 31 March 2005 10:26, Chuck Bunn wrote: Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and I think I figured it out. He's getting the digest version! That's why we think he's on crack. It soesn't make sense having these problems with email. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider problems
On Thursday 31 March 2005 15:59, Kellner, Peter wrote: Ping runs as a low priority service so it is not realistic to measure response time using ping. Try tracepath. It's not using port 7 and can be used by normal users. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium - Asterisk Download Ftp Site link Invalid
On Tuesday 29 March 2005 20:31, Matt Riddell wrote: Kanuri, Seshu (Company IT) wrote: I am going to update WIKI with the latest build number. Yesterday I ended up downloading an old version as WIKI still has 1.0 on the CVS page. Downloading -r1-0 will download the latest in the 1.0 branch (stable). The current latest version in this branch is 1.0.7. Therefore -r1-0 should download -r1-0-7 :) Yes, which it does. Of course the problem is that we keep on having non specific version numbers on update. (Unless you specify the full version.) This frequently causes confusions as people try to figure out which version they have. I recommend that you specify the full version you are downloading as that will be written into the .version file. I.e. specify 1-0-7 or 1-0-6 depending on what you are looking for, vs just 1-0. My update script is currently allowing you to specify this in the customizable section. I'm also strongly considering the ability to specify sub version on the command line when you make updates, to make it easier. (Currently it makes a backup of your source files before altering them, using version number as directory name. During this time it uses the version number available and todays date and time.) I don't know how many are reading this thread and are a user of my script (asterisk-update.sh) but if you have a feeling either way please let me know. Since this is not really vital for all to read I suggest sending my your views off list. And I suppose I should say something for those as lazy as me, who prefer to automate things. The script is an extensive collection of functions around installing and updating Asterisk. It's available at szmidt.org/asterisk/asterisk-update.sh Don't forget to chmod it to 700 after downloading it. : ) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
On Sunday 27 March 2005 22:30, Jim Sturtevant wrote: What product from Sangom and at what price point? Thx See original poster below. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Sunday, March 27, 2005 6:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] small qos switch On Sunday 27 March 2005 13:48, Jim Sturtevant wrote: How about considering the linksys WRT54G (approx $59) with SVEASOFT firmware ($29) www.sveasoft.com which provides QOS by port, IP, and/or traffic type plus VPN, SNMP, etc. and WiFi to boot. Maybe because the Sangoma card will run circles around it purely from a performance view, never mind the quality. Which is usually important to a business... You can buy 400 series servers from Dell for around $350, new. Run your firewall (iptables) and NAT on that computer. You can get a Sangoma DSL PCI card for about $115--it has QoS. You'll have professional grade infrastructure for not that much money. What's not elegant about that? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On Monday 28 March 2005 12:19, Jon Walsh wrote: How does one downlaod the upgrade only is there the ability to do so from the software or do you need to re-burn an iso or is the iso an upgrade version or the whole install over again? Jonathan You can do it manually, or through a script like this one: wget szmidt.org/asterisk/asterisk-update.sh There's a line which let's you specify a specific release like 1-0-7. You can get both the developer version or stable. Don't forget to chmod 700 /usr/local/sbin/asterisk-update.sh -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On Monday 28 March 2005 12:32, Robert Webb wrote: Steve, I believe you have confused AMP-1.10.007 with ASterisk 1.0.7. This was an inquiry about AMP and not Asterisk. Two different beasts. Hehe, I do believe you are right! Thanks! Maybe I should practice my reading... : ) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First second choppy
On Monday 28 March 2005 19:02, Kevin P. Fleming wrote: Robert Goodyear wrote: Anyone know if WAIT is not advisable to workaround the problem Noah's asking about? I always Wait(1) before answering an incoming PSTN or SIP call; there are just too many cases where the media path is not quite ready to start sending audio. There is no need to using Ringing(); that should already be in place via whatever means sent you the call in the first place. Yes, I wait usually for 2 seconds for the same reasons. Plus people expect to hear at least one ring. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] -Using Reply To on Asterisk List-
Hi all, With todays modern email clients, email threads are kept in a separate header. In other words it has nothing to do with the Subject line. When you press Reply To and then change the subject all you end up doing is hijacking someone elses thread. Which is very annoying to the rest of us. To avoid this instead right click on the asterisk-users name and select Send To or New, to start a new thread. This will stop people from being annoyed with you, which may actually result in less responses to help you. Thank you for your interest in Asterisk, good luck! -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
On Sunday 27 March 2005 13:48, Jim Sturtevant wrote: How about considering the linksys WRT54G (approx $59) with SVEASOFT firmware ($29) www.sveasoft.com which provides QOS by port, IP, and/or traffic type plus VPN, SNMP, etc. and WiFi to boot. Maybe because the Sangoma card will run circles around it purely from a performance view, never mind the quality. Which is usually important to a business... You can buy 400 series servers from Dell for around $350, new. Run your firewall (iptables) and NAT on that computer. You can get a Sangoma DSL PCI card for about $115--it has QoS. You'll have professional grade infrastructure for not that much money. What's not elegant about that? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Forwarding to regular numbers?
On Friday 25 March 2005 10:09, Jeff R Glassman wrote: Message: 16 Date: Fri, 25 Mar 2005 01:06:21 -0700 From: JD [EMAIL PROTECTED] Subject: [Asterisk-Users] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I'm trying to set up extensions and have them forward to my cell phone or other phones I have and include them in call groups. I tried *72480204 and *7298480204, I get the recording that unconditional forwarding is set to that number.. but when I call that extension I just get silence and eventually it hangs up. Someone throw me a clue stick? Hmm, I'm not sure if I get what you are asking for correctly but this is what I do. One line dials my extension with a timout value like 20 sec. One could put a message saying trying next extension, please wait. Next line dials my cell with a timeout value like 20 sec. Last line (timeout) goes to VM. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup for linux/asterisk
On Friday 25 March 2005 20:44, Tzafrir Cohen wrote: One nice system-backup program is mondo-rescue. It provides a complete system backup. It can build a bootable rescue CD (actually: ISO images of such a CD) which automates the recovery process even for multiple partitions. The target can be cd images, tape, remote nfs partition, local files, or whatever. http://www.mondorescue.org/ Yes, very practical! Can also be integrated into your very own custom script that does whatever you need. I forgot the details, but a friend stuck with windoze told me about these nice features his new and expensive backup s/w has. I was smiling to myself because I had already been doing those things with a shell script I wrote. True I don't have a fancy GUI but I could have it all run in ncurses if that was needed. Mondo restores your disk very nicely too. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk E911?
On Wednesday 16 March 2005 16:34, Kevin P. Fleming wrote: Wolfgang S. Rupprecht wrote: Have the spa3k use an S0 dialplan: Come On Guys! Get your own thread! I've never had a problem with calling 911 and reporting the address that goes with that Caller-ID for their database. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Center software opensource or commercial
On Wednesday 16 March 2005 17:27, Harald Milz wrote: Kevin P. Fleming [EMAIL PROTECTED] wrote: promise of the Opteron's potential :-) There are very cheap systems out there (designed for workstations using Athlon64) and there are very good systems out there, and the latter can be had for reasonable prices. Well, as a matter of fact, my sales collegues sell Opteron pizzaboxes for HPC clusters like sliced bread - to the automotive industry. These folks usually can't have enough computing power for the buck. But this is utterly off-topic to asterisk ;-) No it isn't. This is very good information for those looking at hardware to run it on. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom phones
On Thursday 27 January 2005 10:45, J Thomas wrote: As a matter of fact, my current client needs 120 phones to work with Asterisk. I have to make a decision soon about which one do I give to him. Hmm. In business service and quality is more important to me than price. After all I'm buying peace of mind, and happier customer relations. How much is that worth to you? By the way people, please trunkate your replies! The thread is right there for someone to follow, you don't need to leave it all there. Please! -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On Monday 24 January 2005 02:52, Peter Svensson wrote: On Sun, 23 Jan 2005, Andrew Kohlsmith wrote: Why would the heads come in contact with the platters on a powerfail? The arms are very rigid -- the heads only float a few thousandths of an inch Well, I'm sorry but I find this whole discussion on why you should have a UPS a bit silly. Electronics are sensitive to ... electricity. May it come in sudden drops just as the data is only in cache someplace, or pulsing power going on and off and back on. Never mind spikes. Fortunately we have pretty good equipment these days that can handle a lot of abuse. But why would anyone argue against it? Either you have the money for it or not. The chance of loosing equipment is there either way. Buy a good UPS and use it if you can. Period. The days of shoddy UPS's are long gone, unless you always buy the cheapest stuff you can find all the time. In which case you might be able to find something crappy. APC gives good support and make decent UPS's at a decent price. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On Monday 24 January 2005 12:12, Steve Prior wrote: One word of caution in case you have X10 equipment. I recently found out the hard way that some of APC's newest UPS models will cause interference with X10 signals going over the powerline. I'm not talking about the X10 signal not going through the UPC - that would be expected. I'm saying that in my case it interfered with X10 signals elsewhere in the circuit the UPC was on. Plugging the UPC into an X10 noise filter solved the problem. Steve steve szmidt wrote: The days of shoddy UPS's are long gone, unless you always buy the cheapest stuff you can find all the time. In which case you might be able to find something crappy. APC gives good support and make decent UPS's at a decent price. That's interesting. Good fix too! I suspect that might not be all too uncommon as they all generate tones for the frequence. Have you tried it with a few different UPS's? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Correct way to update Asterisk
On Monday 24 January 2005 23:31, Pat Delaney wrote: Pardon the newbie post. I installed Asterisk on a test system using the [EMAIL PROTECTED] cd image. When you boot from [EMAIL PROTECTED] is installs an O/S and Asterisk on your PC. How cool is that. But I was wondering if someone could point me in the right direction for updating the version that I have. I'm new to CVS, how do I determine what version to build? Is there a primer on how to download the latest version and install it? If I manage to figure out how to pull it down, when I build it and install, will it overwrite my configurations? Sorry again for the dumb questions Pat You could try my update script at: szmidt.org/asterisk/asterisk-update.sh It will backup what you have and update, compile and install it for you. You can even do it in a number of different ways. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Status on FOP
On Monday 10 January 2005 04:10 pm, Richard Lyman wrote: Joe Dennick wrote: The hype and documentation for the last couple of releases of the Flash Operator Panel claim that the Panel can be configured to either change the LED for a phone, or the name of a phone to indicate when that phone is logged into a queue. I've tried on two different versions (0.18 and 0.19) on two different systems to get this feature to work, and have been completely unsuccessful. Any hints you can provide would be greatly appreciated. Thank you! Joe Dennick [EMAIL PROTECTED] You may of course join the list for the panel, this being for asterisk and all. [EMAIL PROTECTED] -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how is a upgrade performed?
On Friday 31 December 2004 09:41 pm, Charles S. Antrim wrote: I have a stable server and want to upgrade. How do I upgrade to the latest version of * ? Try (wget or http) szmidt.org/asterisk/asterisk-update.sh Run it without any parameters to see your available options. It will perform a backup first and then whatever you ask. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot transfer after queue agent picks up call
On Sunday 26 December 2004 11:13 am, steve szmidt wrote: I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? Well they can do supervised transfers, but not unsupervised ones. I kind of expected it to be the other way around. Is this normal? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot transfer after queue agent picks up c all
On Sunday 26 December 2004 02:57 pm, Hecken, Guido wrote: I had the same problem with snom 190 phones. Using the transfer with # instead of Transfer Button on the phone worked for me. In my configuration REFER was not send, so the transfer with the button on the phone did not work. Guido Hecken Yes, but I think that's another problem disrelated to queues. Use the softbutton and your transfer will work fine. -Ursprüngliche Nachricht- Von: steve szmidt [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 26. Dezember 2004 17:14 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Cannot transfer after queue agent picks up call I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comdial PBX -- can use Asterisk as VM box?
On Monday 27 December 2004 10:53 am, Andrew Thompson wrote: steve szmidt wrote: If you terminate the T's in the Asterisk box and then put patch cables between the Asterisk box and your Comdial, you can probably accomplish these things. You might need to detect what your Comdial does to talk to a VM system and then configure Asterisk to answer properly. What's the best way to figure this out? I'm looking to replace a VM that talks to a phone system over analog lines and a Dialogic card. I am guessing the phone system rings the voicemail(phone system provides dial tone), but I'm not sure how extensions and digits are being used to make the rest of the features work. Is there an application I can use to listen on a line for flashes, digits, callerid, and did-type info? If you can nail it on the network side it's easy. Ethereal will record and even graph different protocols for you. On the phone side one there are specific tools but they cost usually a lot of money. (Starting over $10K.) If it's coming in over a serial port one can rig something to listen and record that, but you'll probably need to be handy with a breakout box. (In effect it opens up the serial cable so you can configure it the way you want to.) I could not give you how to do that without just figuring out and doing it myself and I'm afraid I don't have that kind of time. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk from CVS
On Friday 10 December 2004 03:41 pm, Eric Wieling aka ManxPower wrote: Adi Linden wrote: I admit that this might be some very basic question... How do I obtain Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3? -r v1-0 will get you the latest 1.0.x CVS. Basically it will get you the latest release (1.0.3) plus any patches that will be in the next release of 1.0.x It sure would be dandy if it showed the full version too, not just CVS-v1-0-(date). Would make it easier to quickly ascertain the version. Is this a cvs issue, not showing the full version? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot transfer with Cisco or Snom
On Tuesday 21 December 2004 10:36 pm, Tracy R Reed wrote: I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about Use the soft button! -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comdial PBX -- can use Asterisk as VM box?
On Tuesday 07 December 2004 04:18 pm, Matt Darnell wrote: On Tue, 7 Dec 2004 12:58:11 -0500, George Herndon [EMAIL PROTECTED] wrote: On Dec 7, 2004, at 8:48 AM, [EMAIL PROTECTED] wrote: ken , i too have a comdial analog pbx. i'm running a seperate vm system and would like to migrate to asterisk. right now, my comdial hands off calls via serial connections to my vm box. i don't really know what i'm talking about, but i'd like to find a solution whereby i could accept the T1s (2 in my case) to an asterisk server, route calls to vm as necessary and then hand station calls out to my existing PBX. some clients could be converted over to new IP phones or software based phones (customer service, if quality is good enough) and some clients would remain analog. if anyone is doing this (or a similar but proven and technically correct workflow) let me know. If you terminate the T's in the Asterisk box and then put patch cables between the Asterisk box and your Comdial, you can probably accomplish these things. You might need to detect what your Comdial does to talk to a VM system and then configure Asterisk to answer properly. If you can make Comdial send calls to a specific number, it can make a call that is received by Asterisk and route it to it's VM. Using the same type of interception you can add VoIP extensions that are accessable from your Comdial. The question is how flexible is Comdial for having it make extension type dialing out to the T1's? If it will send out what you tell it to, then Asterisk can receive those and be an extension of the Comdial box. Calls from Asterisk extensions are easily routed to the correct interface and received by Comdial. So if you can configure routing tables on it you should be OK. Digium has a quad T1 card so you have all done on one card. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot transfer after queue agent picks up call
I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Status of linux 2.6 support
On Friday 03 December 2004 08:13 am, Clint Guillot wrote: I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? Still working just fine. It has worked for many months. The simple way is to just install kernel source and add a symlink ln -s /usr/src/linux /usr/src/linux-2.6 if I'm not mistaken. You can make changes to zaptel (I think it is) and it will work without source but I don't want to have to make changes to that source. This would work around the distro problems with using the new 2.6 header info (which saves you from needing the source). Actually, you can get a script that does it all for you at szmidt.org/asterisk/asterisk-update.sh Just type asterisk-update.sh to get the parameters. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and verizon DSL
On Wednesday 24 November 2004 08:27 pm, Scott Laird wrote: On Nov 24, 2004, at 4:18 PM, [EMAIL PROTECTED] wrote: Is anyone succesfully running Asterisk behind verizon residential DSL? I seem to be having some problems with my Asterisk server switching to Verizon. I'm attempting to do some troubleshooting, but I'm really interested in knowing of anyone's setup that already has Asterisk working with Verizon residential DSL. Mine works okay, but I'm using Verizon's business DSL with a static IP. Scott I have a residential DSL up on a Westel modem in Verizon land. No problem. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot open /dev/dsp
On Thursday 25 November 2004 12:08 am, Norman Zhang wrote: Cannot open /dev/dsp: file or directory not found That means you probably don't have a soundcard configured. I don't have one in my test box either, but that doesn't prohibit asterisk from starting up. it just means you can't do certain things from the CLI. You are right. I don't have a sound card in this box. It's suppose to be PBX. ALSA is started though. Try starting up asterisk in verbose mode, a-la: asterisk -vvvc Are all those v's for real? Asterisk supports 9 levels of verbosity. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
On Sunday 21 November 2004 11:42 am, Gregory Junker wrote: Not all over $500 - a quick search finds: For purposes of replacing a receptionist console with a touch screen (for example, replacing a 6x9 grid of buttons), that would be too small as well. Greg Another strong possibility is that after a while, few operators would be willing to continue holding their arms in the air to operate a touch screen. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
On Sunday 21 November 2004 11:50 am, Gregory Junker wrote: Another strong possibility is that after a while, few operators would be willing to continue holding their arms in the air to operate a touch screen. Why would they be holding their arms in the air? You mount the touch panel in the same place at the same angle as the current console... Greg Yes, that would be a better way. But that also requires the available desktop space to do that. Either way I'd make sure operators were happy with that before investing all my beans into it. A keyboard is usually a better all round solution. If not as fun. I investigated doing the same a while back, but ran into these issues that makes it less workable. In the end I decided against it. But I'd love to see someone being successful with it as it looks snazzier. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI 0x3301 Problem
On Thursday 18 November 2004 04:01 am, Sergio Serrano wrote: Hi all, I have a PBX working for a year with an Eicon Diva Server 4BRI. One day it was a storm and nothing occurs, but after a a few days I can't send and receive any calls. I have connected TEIs to Asterisk and other PBX and when I try to dial, I hear correct tone two times, but then line hangup, with the next trace: Hi, Please don't start a new thread in someone elses thread. I know you changed the subject line, but that does not remove the old thread, as that information is stored in the header of the email. The correct way to save keystrokes is not to change the subject but to right click on the list name and selecting New. Thanks, -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_icd compile problem
On Thursday 18 November 2004 07:49 am, Sergio Serrano wrote: Hi all, I try to compile app_icd to test it but I can't compile it. I have installed asterisk 1.0.2 and I download ICD and put files into /usr/src/asterisk/apps/icd directory. I think that make.conf in icd directory is ok but when I try to compile icd I obtain next error: === Compile: /usr/src/asterisk/apps/icd/app_icd.c (app_icd.o) app_icd.c:66: Hi, Please don't start a new thread in someone elses thread. I know you changed the subject line, but that does not remove the old thread, as that information is stored in the header of the email. The correct way to save keystrokes is not to change the subject but to right click on the list name and selecting New. Thanks, -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting - are we there yet?
On Tuesday 16 November 2004 12:46 pm, Steven Critchfield wrote: On Tue, 2004-11-16 at 11:12 -0600, Jay Milk wrote: I'm a fairly reasonable person, and I have yet to see one good argument (and quoting netiquette is not on argument, that's opinion) for bottom-posting. To me, it is terribly inefficient and wastes time, especially when you hide your post between the original message and some ludicrously elaborate signature. Top-posting, to me, is more logical, as it presents the answer in a prominent position. And inline-posting makes sense when you're responding to multiple questions or points in an email... But you are under the sometimes false assumption that your answer is a) good for just that instance of the question, b) The proper answer without needing further discussion. If your answer is a one off and you are willing to repeat that answer every time question X comes up, then fine, you waste all of our time and bandwidth. Else you answer in a manner to wich someone looking in the archives can follow from the question to the answer and see if it applies. Remember your answer will probably still apply in 1-2 years. As someone who's been online since the beginning of the web I can certainly appreciate Jay's, and other's similar, views. But as Steven points out very well, it's not just about ourselves. We live in a community and the degree of order vs confusion we have is all up to us. Those who add to the confusion, ignorantly, or otherwise, are not helping. I've been tempted to top post many times, but I don't want to set that example for others to follow because I've seen how easy things go awry and how hard it is to get many back onboard. It's not about forcing you to do something against your will. It's about educating each other to understand what their actions do. Top posting means more confusion when others come and try to figure something out. If you want to receive help, are you also willing to contribute back? Then once they have that understanding it becomes a matter of integrity of whether or not they contribute. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation
It is also being used by IBM against SCO. And so if IBM attorneys think it's good, there's good chance it is. Actually my comment was not a serious this is why reply, it was intended as humerous reply to this silly discussion. (As there really is no problem with the soundness of the GPL.) Nor do I give a heck what any layman say on the subject as THAT is the real joke. FSF has designed and used the GPL Very Very effectively without needing to go to court, for years. Against numerous violators. Which is really the way it should be. (So tight that other lawyers don't even bother to challange it in court.) It was designed by some very competent license attorneys and has been acknowledged as a very good license by other outstanding attorneys. Of which I don't see one single one on this list. Listening to discussions about law by people who are not layers, which at that would be practicing in the appropriate areas, is like townspeople getting together shooting the shit. It's keeps them busy and entertained, and sometimes rallied up over nothing. Since this is an area which seem to keep peoples imagination going on forever maybe someone should start a small server (Yahoo offers this) to discuss the GPL. Should be very busy and entertaining. Having long since gotten bored with this thread I only dipped in to indicate the futility of this discussion. The thread started out with a honest attempt to put attention on someone that appeared to be GPL violator. Digium put an end to the discussion but the thread refused to die. If someone thinks he has found a valid problem with the GPL why not DO something about it and send off an email to FSF. These discussions can at this point only result in upsetting people who buy into arbitraries conjugated by laymen. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Digium
After seing things readily get out of hand on some subjects I offer this data: How do you bring down a group like Asterisk? You split it up. You create friction and fractions : ) within the group. Now you have the group fighting itself. Anyone who has a valid concern about f.ex. the license Digium has, should take good care in how he spreads that view. I see people who spread this kind of misinformation as a threat to the group. What is your actual intention? And what effects are you creating? If you are continuing creating friction in a group that don't have a problem with the licensing in the first place, then your intentions must be to bring down this group. And even if that's not your intention, that's the direction of your actions. People should be aware that Asterisk DOES pose a real threat, together with the rest of the Open Source, against entrenched businesses. They have a REAL good motivation not to let this cat out of the box. Some people don't really understand what they are doing and help undermining the group by pushing angles and views that breaks up the unity of the group. For example the programmers that are contributing code to Asterisk do so of free will. They have each one of them agreed to the licensing with Digium. If you don't want your code inserted with the main code you don't need it. ANYONE WHO IS NOT CONTRIBUTING CODE BUT WHO IS SPREADING THE WORD ON HOW WRONG SOMETHING ABOUT IT IS, IS UNDERMINING THE GROUP! And most likely, that is their intentions too. This may seem harsh and unfriendly but is nevertheless true. Engaging in a discussion, beyond simply pointing to the FACTS, is aiding such a person. If someone have an honest concern with such issues, they should pursue that in a manner that was not destructive to the group. Why unsettle people just because you don't yet know if you even have a valid point? Get it validated with the proper sources. Check with an attorney and or FSF. Share the result with Digium. Whom, if you found a proper problem, would act to resolve it. If Digium refused to deal with this new problem, then and only then would it be proper to inform the group with ALL the data, so they can educate themselves and see if They care. It should be a CLEAN FACTUAL MESSAGE. Crying generalities or arbitraries does not help anyone as it cannot be acted upon. They are simply destructive or at best a waste of peoples time. A percentage of the population seem bent on being more destructive than helpful. They seem unable to do something without causing more damage than help. We have all seen such people in our lives. Let's not help such people keep a foothold in this group! -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation
On Saturday 13 November 2004 12:19 pm, Martin List-Petersen wrote: It has been tested in city/county court in Munich (Germany) and found valid (http://yro.slashdot.org/article.pl?sid=04/07/23/1558219tid=117), not that that might help anybody in the US, but it is a start. Kind regards, Martin List-Petersen It is also being used by IBM against SCO. And so if IBM attorneys think it's good, there's good chance it is. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT
On Saturday 13 November 2004 01:57 pm, Walter Willis wrote: the asterisk suport NAT as ser? or need modules from modules or special cofiguration? Hmm, your English is a bit too crippled to understand. I'm guessing you are asking if Asterisk supports NAT as something (server?) And then if it needs modules and special configurations. If you are asking if Asterisk can work through NAT then the answer is yes. But the real question is whether or not you are going to use SIP or IAX. IAX does NAT very well whereas SIP is problematic. You need to go to the wiki and read up about Asterisk to use it. It requires a lot of work to understand and yes, you need to configure it. Go to http://www.voip-info.org/wiki-Asterisk and read up on it. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation -- what about IPEYA
On Friday 12 November 2004 08:53 am, Garry Taylor wrote: I have a friend with one of their boxes, as he is having all sorts of problems with it. It is asterisk, no doubt about it. However, they claim to have written there own overlay on top to do the config. via http. And the FXO/FXS card shipped is the TDM400. No code whatsoever is shipped with the box, source or otherwise. Now it is very good that we all get concerned about Asterisk not being violated. But it's up to Digium, who knows who it has sold a commercial license to and not, to take that action. The correct thing to do is to notify Digium and let them decide what if any should be done. We don't even know who is licensed what way, and this is just creating a vigilante movement based on insufficient knowledge. Further, for Digium to present a list of who does have a commercial license would not be in their interest as that can undermine the customers marketing plans and effective use of their license. So chill out and wait to hear from Digium, don't get into the middle of a legal scene you know nothing about. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can some bady help me ???
On Thursday 11 November 2004 04:39 pm, Geoff Nordli wrote: [EMAIL PROTECTED] scribbled on : ok, mathew and other friends I have this package only and I don't now what a have to do with it I repeat im a new linux user I don't now how compile it. I need for start a list of steps to begin or a place where I can get it thanks rodney If Linux is a struggle for you then you may be better of looking at a Live CD type of installation. I haven't tried Xorcom's Rapid installation yet, but it may be worth a try: http://www.xorcom.com/rapid/index.html Geoff Not a bad idea. I just downloaded the SuSe live DVD and it's a very slick system. Easier than anything else out there. I think they also got a Gnome version and KDE version live CD. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SELinux and Asterisk
Hi, With the release of FC3 SELinux is now enabled by default on Fedora. For more details see http://fedora.redhat.com/docs/selinux-faq-fc3/ This is a great method of adding granular security to Linux. As no policies probably exists at this point (for Asterisk) I realize that it's a good idea to start the design of the neccessary policies. However, SELinux is not for the faint of heart, and with the limiting/crippling abilities that it has, I thought it a good idea if we try to poll the efforts. Looking a bit further on this I also realize that a specific security related list is a possible route to take. For nothing but keeping it more accessable and focused. I also realize that we have not really seen much noise from this particular area, but as anyone with security experience can say that does not mean we do not have potential holes. The Best Practice outlook suggests establishing guidelines for any service running on a server. To this end many of us take extra precautions as to limit a possible violation. SELinux brings with it a long needed granular control over each process, and in general makes a server much more secure. Thus making its benefits obvious to anyone who has a server available online. The sheer volume and noise in Users makes it a hard place to conduct such coordination. Being forced to keep up with the volume just to see what might relate to any particular needs and interests is, as you all know, very time consuming. The process and experience of establishing and using various security modules and methods will obviously have it's own share of problems. As it is so different and yet demands particular attention to details, I want to check for interest in creating and working with a security list for Asterisk. One could use the Developer list but I don't think that's really the best place either as it is not related to developing Asterisk. (As it might require coordination with developers too, I have CC'd that list. Where they can put their thought on the subject stricktly from their point of view as developers.) A security list would obviously carry all issues related to securing an Asterisk box, and as such ought to be with digium, but if some issues for some reason makes that undesirable, I might entertain the option of hosting it on a seperate server. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-capacity systems / trouble with Tyan
On Wednesday 10 November 2004 05:25 pm, Tim Jackson wrote: From my experience the Tyan Tiger MPX is a great board. I've never used it with *, but I have been using it as a high volume samba server for over a year and its never even hicupped. 16:24:30 up 197 days, 20:45, 2 users, load average: 0.94, 0.92, 0.89 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Wednesday, November 10, 2004 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] high-capacity systems / trouble with Tyan Hello, I've had a Tyan dual Athlon MP(2800) machine for a year now and have had several lockups for strange reasons on stock redhat kernel and on custom compiled kernel off of Slackware. I've tried every combination of BIOS settings and changed out all assiciated hardware and found the problem: It's the Tyan. I've also had issues with a couple of SCSI RAID cards when I tried using them with the Tyan card. The problems motherboards has is not usually very visible to most applications. However, when you get an app for which interrupt timings are crucial, you'll notice boards that should have no problems, do. I got a system that cost $15,000 a few years back. It's a dual XEON 550 Intel board. You'd think it would do better than a lot of others. In truth it does very poorly with Asterisk. It cannot handle one single call without having a problem. (Since that it has problems with interrupt timing, it might actually work better if I remove one CPU ...) On a different note. Please do not top post, and when you reply cut out redundant parts of the original email. If you have a problem with typing the mailing address and use reply to get around it - use R-Click and select New instead. This avoids starting new threads in others threads. Changing subject does not change the thread as that info is stored inside the email header. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice asterisk patch
On Wednesday 10 November 2004 07:37 pm, Michael Giagnocavo wrote: Which once again brings home the fact that too few people understand security in the first place. Damn straight. Check out the replies on that thread. It's like my posting about a security list. I was wondering if anyone was even going to reply. As it is they are all replying to my email address. Which fine, but there are not very many. What's this? I didn't see anything about it. I'd definitely be interested. It was called SELinux and Asterisk. The reason I replied to this thread was to make make a bit of noise for someone at BV to notice and maybe at least consider improving their patch model. Write to Bruce Schneier at Counterpane and see if he'll doghouse them on his blog :). -Michael Hehe, yes I know him. A no BS, down to earth guy who REALLY knows his stuff. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice asterisk patch
On Wednesday 10 November 2004 09:35 pm, Tom Lahti wrote: At 02:39 PM 11/10/2004, you wrote: In any case, the patch has been positively identified as being genuine. Which one? Anyone who got an email like that? Get the point? :) Holy beating a dead horse, Batman. To some it's a dead horse, unfortunatley for many it aint. No one is suggesting that because person X read and understood the patch that it makes it a fearless install for anyone who receives anything claiming to be the patch. It makes it a fearless install for person X *only*. If person Y wants the same peace-of-mind, he has to read and understand it himself. Well, I wish I had that experience. What I see everyday is people being too lazy to find out for themselves and just following the loudest recommendations. Even if from a known fool, as it appears to be easier than to check it our for yourself. Since (a) asterisk is not Broadvoice's product, (b) Broadvoice does not even officially support asterisk, and (c) asterisk is an open source project, the *only* appropriate action they can take is to email a patch. I guess I don't agree with you on that either as I could see a few different ways. They are a business trying to earn a living on their own products and they did what they did to alleviate their OWN problem and save their OWN network, the rest of it be damned, which is totally appropriate for any for-profit business. They could just as easily have said screw you asterisk people and disabled asterisk's ability to register with their servers and not done anything about asterisk's lack of ability in the registration area. Yes, they could. Still does not make mailing patches an ideal way of doing it. Now, thanks to their effort, we have an improved asterisk with greater ability and compatibility. Since noone else has said it, I'll say Thanks, Broadvoice. We're glad to have you contribute to the asterisk codebase, and good work! You know, I'm sure they are decent people. (Giving them the benefit of doubt since I don't know them.) But surely they made a business decision. Realizing there's a big potential with all these Asterisk people. This does not mean I don't also see it as a thing to appreciate. I think both sides can be right on this one. But it'd be more wrong for not pointing out what IS a bad way of doing it. Unless they are made up by a bunch of insecure kids, they will no doubt take it for what it is, a notice about something they did which is insecure. Too many people are afraid of rocking the boat by speaking up and so sit quietly watching it take in water through a hole. Look at the whole microsft mess. They got most of the computer world in hock over the same issues. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS RPMs for Mandrake 10 (Zaptel and, Asterisk)
On Sunday 07 November 2004 07:39 am, Clive Carter wrote: Dear Scott I am new user of Mandrake 10 And very excited at the idea to work with Asterisk but, as you can imagine. I am currently blocked because of the kernel 2.6.. the Wildcard X100P drivers . I would be more than happy to get test your source RPMs for zaptel and asterisk And so would I !! Don't know what block you are talking about. MDK 10 the 2.6 kernel works great with X100P. (Unless something recently has changed.) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3
On Wednesday 03 November 2004 08:05 am, James Botham wrote: just interested to know what the contents of you /dev/zap/ctl directory are. ctl is not a directory. Look here: crw-r--r-- 1 root root 196, 0 Oct 30 16:40 /dev/zap/ctl And no you cannot readily cat a dev. I've not had to create any device files for a long time so I've forgotten how you do it, but the c means its a character device. It's a device that is read one character at a time. The command should be mknod so a man mknod should give you the details. And I cannot imagine why it was not created for you. It seems plausable that it should have been created at least by the time you run the zaptel install script. You could also just reinstall the binaries. There's script on szmidt.org/asterisk/asterisk-update.sh that will create backups of your source code and download, compile, install the asterisk system, etc. Simply run it without any parameter and it will give you the syntax. Not that running the install by hand is very hard... but I get lazy and try to automate everything. Which can make things overly complex. So it may still have some bug in it. Though it seems to run just fine now. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom200 - asterisk dtmf (rfc2833)
On Monday 01 November 2004 05:20 am, Arsen Chaloyan wrote: Hello all. Seems, snom200 with 3.x software versions isn't compliant to rfc2833, while no such problem with 2.04.g. Timestamp and event duration are increased simultaneously, while timestamp should points to the beginning of the event and event duration extends forwards from that time. In any case, asterisk is able to detect dtmfs correctly. But I believe that in scenarios where asterisk acts as IP-PSTN gateway the dtmf sequence generated by asterisk in PSTN side can be incorrect. In other words dtmfs can be overlapped if they quickly pressed on snom. Unfortunately, I have no PSTN card installed on my asterisk. Can somebody check if this works? snom200(2.04.g vs 3.x) --ip-- asterisk --pstn-- phone Hmm, I don't see that as even being an issue as snom only sends the digits once you press OK (except when asterisk is configured to read it live). If you mean dtmf after a connection is established, then that's true for any equipment. I.e. a too brief signal is always possible. I'd call that user error. Using my snom I can't say I've noticed anything like the above during normal use. Now there is another problem in that sometimes I don't get it to dial out. I get a period of silence after hitting OK. A while later I get a dialtone. Or I end up with a fast busy. But I _think_ that is related to the non-digium hw I'm testing for pstn access. If I press redial a number of times I'll get through. My thought is that if it was a snom error then redial would never work, as it's only repeating what I typed. And, it behaves the same way if I manually redial a number of times. My firmware is V3.52. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List Issue
As I see it we are all a bit better and a bit worse on various subjects. But one thing most of us have in common is being basically decent people trying to help. Very seldom do we get some fool who's trying to undermine or sabotage our efforts. One thing that helps keeping a list friendly is to keep demeaning comments off-list, and only after several failures do so publicly. Even then it's seldom of any value, as few others usually even want to read it. 97.5% of the time it's only interesting to two people. Being that the size of our list has so many different people with various backgrounds and levels of understanding, not only of Asterisk but also of English, we are bound to run into things that looks, well... bad. It's very easy to read into things the wrong meaning when you don't know the spirit it said in. When I see various things like html or mispostings I always try to handle it offline. (Not that I have not made mistakes.) But it takes the edge of the blade and keeps it a more friendly place. If you feel a guy is full of it tell him offline. And if you ask what he meant in a way that is non confrontational he's more likely to answer properly and realizing the errors of his way, apologize. When a service company makes a mistake with my account, I call them up and make them my ally against the problem that occurred. Makes them willing to solve it. I might even say (in a calm manner) that I'm very unhappy, and add that I know You did not do it, but I want you to know that I'm unhappy. Amazing how far you can get with some elbow grease. (No crude comments here : ) On the all, I think this is a great list with some very competent and willing people. People who's making the growth of Asterisk possible through their valiant efforts. Keep up the good work! -- Steve Szmidt There's always two sides to any dispute, your job is to fully understand them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is NuFone messing up for anybody else?
On Friday 29 October 2004 05:09 pm, Paul Rodan wrote: 1-way audio problems. At least I think it's one way. We hear the remote party breaking up. So it would be NuFone's ability to transmit, upload, or our download bandwidth. We're not having bandwidth issues, we have 4 DS3's at only about 70% of total capacity. I'm having the same issues. Though my call volume is really low. Using other servers than NuFone's I've not had the problem. There has been numerous problems making calls - with no line available. They say it's not them, but it does not help me if the route to them is to laggy or whatever the problem might be. Calling via a server (coast to coast) and I never have a problem, but often I do through NuFone. I know they often work late helping people but I've started a few threads that never went anywhere so I'm in a bit of a mystery as to what is going on. Does not make for a safe business plan. (I use a dedicated WAN pipe with QoS set on my Asterisk box.) Last week I upgraded from a CVS Head from around 9-15-04 to the CVS Stable on 10-26; However, no immediate problems were detected. Smooth upgrade. We place many calls a day and only today is it worst than usual. Just wanted to see if anybody else detected it, apparently not. Will look for another U.S termination provider with similar or better rates and move NuFone to secondary. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Chaney Sent: Friday, October 29, 2004 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is NuFone messing up for anybody else? Can you be more descriptive on what's happening? I use NuFone and haven't had any issues, but I don't make that many calls. On Fri, 29 Oct 2004 14:39:57 -0400, Paul Rodan [EMAIL PROTECTED] wrote: We've been using NuFone for about 2 months, pushing an average of 10,000 minutes a month of Long distance. There have been minor quality issues in the past, maybe to one area code or another. However, I've noticed today more problems than usual. Gotten several calls. Has anybody else noticed this? FYI. I'm running Asterisk CVS-v1-0-10/26/04-07:28:01 on Asterisk Server A, and Asterisk CVS-v1-0-10/25/04-16:04:29 on Asterisk Server B My phone connects to Asterisk Server B. Asterisk Server B connects to Asterisk Server A via IAX, which is off of the same switch. Asterisk Server A connects to NuFone via IAX. Cisco_7960 -SIP- Asterisk_B -IAX- Asterisk_A -IAX- NuFone via Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 190/220
On Friday 29 October 2004 12:32 pm, Ronald Hartmann wrote: Good Day list, I have spent better part of the morning reading through the user group messages and have found some people stating that they are able to get the Transfer Button to work on the Snom 190/220 Yes, press the softbutton that says Xfer (means transfer). -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xorcom Rapid Asterisk distro beta 0.5.2
On Sunday 10 October 2004 06:41 am, Tzafrir Cohen wrote: Hi folks Hello to all, We have created a simple Debian-based distribution of Asterisk. A CD image of an installer(150MB, requires no extra packages from the 'net) that installs Debian and Asterisk simple and easy. You are invited to take a look at: http://www.xorcom.com/rapid/ The image is free as in GPL. Sources included on the image. Any comments will be appreciated, either via the website or directly to me. I'd like to thank all the users and developers who helped me on #asterisk , #debian-boot and other places. Being posed as an Asterisk distro I decided to reply to the list. This is a nice and fast install ending up using the whole of 334M on a single partition. I used an old 600 MHz machine with 256M RAM and it went pretty fast and smooth. Though I can't for the life of me understand why it defaults to having these ports open by default: porttcp udp service 9 x x discard 13 x daytime 37 x time 2000x callbook I know I don't want to offer any of these services to the Internet. 9/13/37 are never used these days as those services were found too easy to hack through. That was a number of years ago and of course they could be improved. But still does not explain why they are open. My SIP devices uses 123. Port 2000 has been reported as recently as the 25th Oct to be an increasing new IIS PCT exploit. One usually prefer to keep a low profile with servers. This one is asking for attention. To their defense, if you read the release notes, they do recommend against using this in a production environment. I'd like to see a more prominent warning. And during the ever simple install it does not verify the root password. You better know what you type. It does not have ssh installed. Not being a debian user I'm not sure if there's a good reason to not include ssh in the default install. Except to keep things to bare bones. Though I would be hard to not have space for ssh. The game Banner could be skipped if space is the target. All in all it has lots of tools linked through a menu system that works pretty decently. Plenty enough for a server. I guess having an ability to edit asterisk from there could be added. Otherwise it's quite complete. I managed to install ssh, and mc, easily enough (from the CD I think, it seems too fast to have come down over the net). Somehow I've managed to make this my first direct contact with building a Debian system. It would be VERY hard to make it any easier. The one thing I'd like to see is a menu option that opens the services I need After the install. Not open by default. Asterisk from 05/31/04 is running on kernel 2.4.27. There's a minor point of having a broken vm link in /var/spool/asterisk. Having said all that, I think they have done a great job of creating a single Asterisk CD. Some honest work went into getting this done. As a contribution to Asterisk I think it's a very good thing! If the next release continous this well, it should be a very popular distro for our community! -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gentoo
On Tuesday 26 October 2004 01:11 pm, David Ishmael wrote: I'm planning on installing Gentoo as my distro and wanted to check here to see if anyone has any tips or ticks I should know about that aren't on the Wiki. I'm installing the TDM400P with a single FXO module for connectivity to my home PSTN connection. Anything I should know or do that would make this go easy? Besides from easy, there's one consideration to keep in mind. How good are you on locking down a linux box? Just using ipfilter/shorewall does not make a secure box. (Mandrake 10 does a good job of locking down a box. Minus whatever You install and run that might be a liability.) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfering Calls
On Monday 25 October 2004 04:43 pm, [EMAIL PROTECTED] wrote: I have tried that on the GrandStream Budgetone phones and the transfer does not work on them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, October 25, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfering Calls Brian J. Rathman wrote: I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. The t and T options to Dial() provide # transfers. Use the transfer function of your SIP phone and don't use # transfers. I think the problem is the version of firmware. I used to be able to transfer fine but now I can't. On the other hand a lag I used to have is now gone. My V is 3.52. Using the hard Transfer button don't work, but using the softbutton does. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: cannot call Grandstream
On Friday 22 October 2004 02:05 pm, Neil Cherry wrote: David Ishmael wrote: I think my Netgear router will try to lease the same DHCP address to a device based on MAC automatically each time the device queries for an address (but I'm not 100% sure about that, never really watched it). So the problem is with the address changing? I can't infer that from the 2 examples as it may be some other problem with the DHCP implementation on the DHCP server. Though it may be a possibility. I like to have the stationary IP devices to have a permanent IP address. It just makes it easier to admin my local DNS (I have too many devices to remember all the IP addresses). Hmmm. In my opinion DHCP is mostly a false time saver anyway. It's true you can just plug in a host and have it get an ip nice and easily. But I prefer to know who's IP is on the wire with a minimum of fuss. I like to be able to notice that nnn is being involved far too often in that XYZ problem, or whatever. Plus it's one less service to maintain. Whenever I add a host I spend a little more time with configuring it but that's better than chasing leases as far as I'm concerned. Eases LAN maintenance a lot. True, as an ISP I would use DHCP. It's quite suitable there as I would have more limited resources. But on a LAN it's hard to run out of IP's. It's kind of how windows got popular, thanks to the apparent easier way of doing things, and how lazy we all seem to be. Anyway, this is on th edge of the topic so I'll stop here. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cheap gig switch? smc, netgear, or 3com?
On Wednesday 20 October 2004 04:47 pm, Matt Hess wrote: Remember, you pay for what you get.. especially with Dell networking equipment. I have heard about several groups who tried the dell switches only to give up on them because the dell switches just didn't perform. Yes, price-wise they look good.. but as far as performance goes.. (that is assuming you want high/solid performance) I'd look elsewhere. Jup, I've read the same. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cheap gig switch? smc, netgear, or 3com?
On Wednesday 20 October 2004 04:08 am, Jay Wilton wrote: Hello, The Smc 8508T goes for about $95, jumbo frame support, lifetime warranty but no QOS. The Netgear GS608 is $ 100, no jumbo frames, 1 year warranty, QOS, gig latency 10U max. The 3com switch reviews that I read were not happy. Does anyone hate or love their home switch? I doubt the jumbo frame support would help voip traffic, but it seems like it wouldn't hurt. I was planning on doing the QOS on linux. Gig support is wanted for file transfers and the future. Thanks to all you nice asterisk people and a few of the mean ones. Jay Haha, a few of the mean ones! I love it! : ) I prefer managed switches but they are all so pricey. The thing to go for with any switch for VoIP use is the ability to deal with QoS. Most of the routers are configured to support it and it does work. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cheap gig switch? smc, netgear, or 3com?
On Thursday 21 October 2004 09:16 am, Matt Hess wrote: There was a thread on NANOG a while back about dell switches and the opinion at the time seemed almost in complete agreement - dell switches stink for everything but pure ipv4 shuffle packets.. unmanaged without any features. They are not ciscos at all.. they have a cisco like interface but then again so does zebra.. but that doesn't make it a cisco either. And imho, being the 'wal-mart' of something isn't necessarily a good thing.. even wal-mart sells some total junk (to put it lightly). And except for only the largest routers, Cisco is overpriced and under powered. Great support but poor value. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on OS Choice
On Thursday 14 October 2004 09:10 am, Alex Barnes wrote: Hi all, I am currently trying to decide what Operating System is best to go for on a customer site. Server will only be running Asterisk / MySQL / Apache / PHP but nothing else. I have only tested Asterisk on SLES 8.1 however I do have experience with RedHat 9 as well. One very good distro for these kinds of setup is Mandrake 10. It has a very easy to configure security setup which will harden the box for you. It's based on RH and is fully compatible with mainstream configuration tools. (Versus f.ex. SuSE) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: {SPAM?} [Asterisk-Users] Asterisk VIA SSH Tunnels
On Thursday 14 October 2004 03:04 pm, Geoff Nordli wrote: [EMAIL PROTECTED] wrote: On Thu, 14 Oct 2004 07:13:04 -0700, Geoff Nordli [EMAIL PROTECTED] wrote: OpenVPN runs on: Linux, Windows 2000/XP and higher, OpenBSD, FreeBSD, NetBSD, Mac OS X, and Solaris. And how many routers and firewalls out there do support OpenVPN? Do Cisco routers support it? On the other hand, IPsec works on all the platforms you mentioned *plus* most routers/firewalls from Linksys toyz up to Cisco and Checkpoint etc etc etc. rgds benjk No argument here. If you want to do gateway to gateway then IPSEC is a solid choice. They pretty much run flawlessly. The only thing I don't like is the kernel modification required on the 2.4 kernel series to embed Openswan/Freeswan into the kernel. Just one more thing to worry about if you need to upgrade the kernel. Since the guy was talking about using an SSH session I assumed he was looking at client to gateway options. IPSEC is not a great option there. An easier solution is to use something like PPTP, but sometimes GRE is not Please don't use PPTP as a security solution, because it really isn't. It's so flawed you can even connect to it without having ANY encryption. Microsoft with their never ending wisdom have incorporated design flaws that make cryptographers and security professionals distrust it, and recommend against its use. Or as the writers of Building Linux Virtual Private Networks says: We recognize that there are times when you must support PPTP ... In either of these cases, we offer our deepest sympathies. supported on every firewall. Plus PPTP requires modification to the ppp kernel modules to support mschap-v2 -- this is also a pain. So something like OpenVPN is a good solution. Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing out with SIP phone problem
On Wednesday 13 October 2004 07:15 am, James Bean wrote:
I am trying to setup a SNOM 190 with my asterisk box but having a few
problems
When a call comes in it connects and rings and I can talk no problems...
If I try to call out with the phone I get...
NOTICE[-165364816]: chan_sip.c:7561 handle_request: Unknown SIP command
'PUBLISH' from '192.168.69.250'
This message does not affect anything. It's a bug with Snom firmware that it
sends out a publish even though you tell it not to. Meanwhile no need to
worry. One of these days Snom will fix it and we won't see the message.
I know dialing out works correctly from my analog phone plugged into my
TDM400P but the sip phone doesn't seem to dial properly?
I updated the latest firmware on the snom190...
The configuration on the SNOM190 is pretty standard with just Line 1
configured for asterisk with the correct password etc, I get the
-- Saved useragent snom190-3.54 for peer snom-james
And
[2]24/12/2001 11:00:09: Registered at registrar as
[EMAIL PROTECTED]
So the phone and asterisk sync and talk ok.
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
srvlookup=no
[snom-james]
type=friend
secret=password removed
host=dynamic
callerid=James 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=900
[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=rfc2833
mailbox=901
/etc/asterisk/extension.conf
[pstn]
exten = s,1,Wait(2)
exten = s,2,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,3,Dial(SIP/snom-james,45,t) ;Dial the group=1 zap card mod
above
exten = s,4,Hangup
;exten = s,5,VoiceMail(u100);Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
include = outgoing
include = voip
include = sip
[outgoing]
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup
[voip]
exten = _1XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 1xx extension to Salisbury
exten = _2XX,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM})
; 2xx extension to Marcoola
exten = 610,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM}) ; 610
to Jindalee
exten = 620,1,Dial(OH323/[EMAIL PROTECTED]/${CALLERIDNUM}) ; 620
to Batteryhill
;exten = _54XX,1,Dial(OH323/[EMAIL PROTECTED]) ; 54 to
Marcoola
;exten = _0754XX,1,Dial(OH323/[EMAIL PROTECTED]); 54 to
Marcoola
[sip]
exten = 690,1,Dial(SIP/snom-james,30,tr)
exten = 690,2,voicemail2,u900
exten = 690,102,voicemail2,b900
exten = 691,1,Dial(SIP/bt-karen,30,tr)
exten = 691,2,voicemail2,u901
exten = 691,102,voicemail,b901
-
Although something strange, on bootup asterisk console displays
WARNING[-165811280]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Non-critical Request)
Any help would be very much appreciated.
James
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--
Steve Szmidt
They that would give up essential liberty for temporary safety
deserve neither liberty nor safety.
Benjamin Franklin
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Re: [Asterisk-Users] Free G.729 ready for download
There's another legal side to all of this which we need to evaluate carefully. Putting the list and Digium, at risk, by being in a position of having it used to break the law. Starting a few years ago ISPs became liable for harboring lawbreaking customers, and ended up answering to the court. If a court can be convinced that a particular list is used to spread illegal copies of let's say G729, then it's possible it could be held liable. The only thing I see missing from those types of court cases at this point, is Digium have probably not received a letter saying their customers are using their resources to violate someones copyright/patent - with a cease and desist letter. So the question is - do we really want to take that chance? Lawers do what they do as that is their livelyhood. If we get someones attention once, it will be that much closer to a second time. The law breaking would be trafficing in illegal copies of G729 with the intent of breaking the law. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Case studies for 120 simultaneous calls on IVR
On Friday 24 September 2004 12:46 pm, Christian Victor wrote: Hi Rgis, Were going to build an IVR system with a TE405P and 4 E1. Were sure that the 120 channels will be filled by 120 simultaneous calls during peak, so we want to have the good server to manage this. We wonder a lot of things and maybe you could help us. - Are you ever build a similar system ? - Does linux use the advantage of Xeon processor ? so we must buy Xeon ? It does. Put I would prefer two single P4 boxes over one dual Xeon. BTW, Asterisk does utilize a dual processor very well. Whereas two computers offer redundancy. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0 Libs
On Friday 24 September 2004 01:53 pm, Anton Tinchev wrote: Whicch version of zaptel and Zapata should I use with 1.0? One should always try to use the same version. CVS will give you all the files you need. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 13 sec. delay what is causing it?
On Saturday 18 September 2004 06:21 pm, Lyle Giese wrote: Perfectly normal. On analog lines, the caller id is set between the 1st and 2nd rings. So Asterisk has to wait for the caller id and depending on the speed of the computer that hosts Asterisk, 13 seconds is exactly right. A normal ring cycle is 2 secs ring on 4 seconds of silence, so the 2nd ring is 12 seconds into the call. I just put in a nice Asus motherboard with a 500 mhz front side bus, 2.4 gig AMD processor 512 meg ram for the pbx here and I get the first ring on the extensions at the same time as the second ring on the incoming ring. I was testing and trialing on a celeron 1.4ghz machine with 256 meg ram and the video borrowed some of the system ram. The analog extensions were not ringing until the third incoming ring on that slow machine. Lyle I've been using 2 seconds for about a year without ever having noticed a missing CID (on analog lines). -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Linux 2.6 Kernel
On Sunday 19 September 2004 10:28 pm, C Wegrzyn wrote: I ran the LiveCD version of Asterisk on my hardware and it worked. I am trying to run it natively on a 2.6 kernel (Gentoo distro), but it keeps getting a seg fault using the sample configuration files. Does Asterisk not work with the 2.6.8 kernel? TIA Chuck Wegrzyn I've been running 2.6 for months using MDK 10.0, on two machines. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon pictures
On 21 Sep 2004 at 15:40, Kristian Kielhofner wrote: Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! I missed the outcome of the dialog about some kind of broadcast from the event, how did that end? -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hey admin: Do we have to have a 92-char reply-to header?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 27 August 2004 12:32 am, Brian Capouch wrote: I don't know who else may be suffering from this, but the ultra-long Reply-to: header seems to break my mail reader. I have been suffering the zanies for the last week or so--mainly showing up as the scrollbar disappearing off the right side of my mail window. Tonight I figured out that it's due to the browser reacting to fit the length of the header. The fix was to stretch my mail window out to about 24, occupying my whole screen. This is Mozilla 1.7/Linux, Slackware 9.0. Thanks. B. This is a Mozilla bug. If you can report it to Mozilla. I use 21 screens running at 1600x1200, the point being that I could not imagine NOT using the whole screen for my email client. I want the one-view see-it-all, view. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBL2AbljK16xgETzkRAsPkAKCIzJYnrJplEMpca8FFt7ecMdpKkQCfXmA4 Byz9F8Pj12UgO8jYCUXfU7Y= =o+3I -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Install on Kernel 2.6.x
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 23 August 2004 03:14 pm, Shawn Parker wrote: i know asterisk itself will install on a linux kernel 2.6.x, but i've seen places say that the zaptel drivers wont? is this still true? is it possible to build asterisk/zaptel on a linux 2.6.x kernel? A number of us already use 2.6. Remember the wiki is your friend. http://voip-info.org/tiki-index.php?page=Asterisk - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBK26oljK16xgETzkRAvctAJ9c3WV0NE5dmI7FhoqKSYprxHIzqACfTJu5 ApNofNxmHUGasGqAl9Lydg4= =YkYZ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telemarketer screening
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 24 August 2004 01:34 am, david kwok wrote: I have been bugging by a telemarketer who does not take any cue at all. So I look up the Asterisk Handbook and send his call with the respect caller id to my voicemail. Has any one implemented any of this feature with database for more caller ids to be included?? David Kwok Be interested, get all the contact information you can. Ask for supervisor. Inform person that you have all contact info and next time they call you will file suit. Ask to be put on their legally reuired do-no-call list. Should be end of it. Or go to small claim court for an easy payoff. Law is very strict on these matters now. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBK3m9ljK16xgETzkRAkzIAKCunhEdA0iHqbyGjVapXGNnGP4kmACfW6a7 pIphiH5N3J4z9/pNKrSe4bY= =+5Kj -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge 750 rackmount
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 24 August 2004 01:02 pm, Steven Critchfield wrote: The big thing to look into is what PCI busses the machine supports. We where very surprised with our Dell when it came with a PCIX slot and a 66mhz 64bit slot. The included ethernet card wasn't directly supported by our install disks and we couldn't install a cheap card to get the install bootstrapped due to the incompatible slots. So after building a special boot disk, all was better except the T100P card I owned wouldn't work in it again due to slot incompatibility. That prompted our TE410P card purchase. After using various Dell's for a few years I'm very weary if I need something not bland. They are very very good at cutting corners at all sorts of places. Their intended public is mostly run of the mill machines. They are the pro's in cutting pennies and turn it into BIG savings. Too many times have I discovered something odd or less than I would have expected. SuperMicro on the other hand have striken a good balance of price and quality. They are really are going after the market with some very solid hardware and not being the cheapest either. Which is really less important in a business environment anyway. Support and quality being more important. It's really true that you get what you pay for. If you can afford quality it's cheaper in the long run. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBK45fljK16xgETzkRAtG+AJ9dStuHpzMSxiQFafCD1SSToRF+TACgknUl yL4OmWEaSPs+abhFM3i1S2Y= =lIxc -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMP Performance
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 24 August 2004 04:00 pm, Tim Jackson wrote: We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? I'm testing a dual Xeon running 650MHz 768MB and I estimate that it might handle about 12 connections on g.729. No SMP issues I'm aware of. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBK8pRljK16xgETzkRAl7kAKDaOZCElZ2oZZgmnyGc2BlM5sjVfgCgzwSP alRl1n7b91TBC2MHah+1TjI= =E06r -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Saturday 21 August 2004 05:22 pm, Andrew Kohlsmith wrote: On Saturday 21 August 2004 14:47, Ed Devine wrote: Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Supermicro server system using software RAID1 on two 9.1G UW2 drives. No issues thus far. ~3 months running. Yeah, I've run s/w radi for long time and never really had any problems. I just replace the drive when one goes bad. Since I always put a spare it works seemlessly. I just have never done it with Asterisk. I don't think I would use s/w raid with it either. I don't want my cpu's to be busy with drives if it's not needed. SCSI then becomes a good choice too. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJ8C8ljK16xgETzkRArLWAKDPm/nV75BEnYGhpZXZi160vJ9jGwCfVqD5 fn+85yuBLyAdCIDnyFjeVgM= =2txp -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound IAX2 calls has no music on hold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 20 August 2004 02:25 pm, Eric Wieling wrote: On Fri, 2004-08-20 at 12:52, Steve Szmidt wrote: Does ANYONE have music on hold working across IAX2? Google does not return anything on the subject. Except I did see on the release notes for 0.7.0 Better support for MOH in IAX2 Yes. It works fine with no special config required. Well, I'm glad it can work. I've never heard any MOH when the call comes in on IAX2. It would be nice to know what the condition is that stops it on IAX2 alone. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJ8HLljK16xgETzkRAi7PAJwIl58TM6nuqLR2EXiWivhGuw4YUwCg4X0G y0jlQqIfWhcWQt0rsFILMxU= =trnX -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 20 August 2004 06:02 am, Thomas Kuepper wrote: use ssh instead of telnet. telnet is a bad idea. And the reason telnet is a bad idea, is because it sends the password in clear text. Today there's no valid reason to use telnet over ssh. Am 20.08.2004 um 11:39 schrieb neil: Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks Neil -- Thomas Küpper 01063 Telecom GmbH Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: [EMAIL PROTECTED] E-Mail: [EMAIL PROTECTED] Homepage: http://www.01063telecom.de --- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --- This message is confidential and may be privileged. It is intended solely for the named addressee. If you are not the intended recipient please inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of the information contained in this message, the statements set forth above are not legally binding. In connection therewith, we also refer to our governing regulations of concerning signatory authority published in the standard bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --- - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJfVpljK16xgETzkRApNZAJ4vGi82IvIvMv+8eQidWUY5TNj0pACgiDCL 1W8yYs8b46mj+YZxvO7kHMI= =I02Y -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 20 August 2004 12:22 pm, Chris Shaw wrote: LOL it was so long ago, I didn't think about that reason... :) - Original Message - From: Walt Reed [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 9:13 AM Subject: Re: [Asterisk-Users] telnet and Root On Fri, Aug 20, 2004 at 08:40:43AM -0700, Chris Shaw said: ...Today there's no valid reason to use telnet over ssh. Was there ever a valid reason? Maybe export restrictions on crypto? I've never EVER used telnet or rlogin, SSH is so much nicer anyway... Yeah. Some of us were around before ssh existed. :-) Plus it's still a good tool for talking to various services like a mailserver to debug connections. (You can specify the port to connect to.) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJi4OljK16xgETzkRArQqAKChzn/TaSXGAbus4aX9gt4j5vY9JwCg2pA0 0lFshbWX7dzFVZXCTOVSUkA= =UfTn -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 20 August 2004 01:14 pm, Steven Critchfield wrote: On Fri, 2004-08-20 at 11:59, Steve Szmidt wrote: - Original Message - From: Walt Reed [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 9:13 AM Subject: Re: [Asterisk-Users] telnet and Root On Fri, Aug 20, 2004 at 08:40:43AM -0700, Chris Shaw said: ...Today there's no valid reason to use telnet over ssh. Was there ever a valid reason? Maybe export restrictions on crypto? I've never EVER used telnet or rlogin, SSH is so much nicer anyway... Yeah. Some of us were around before ssh existed. :-) Plus it's still a good tool for talking to various services like a mailserver to debug connections. (You can specify the port to connect to.) Outside of SMTP and www, what are you doing with a open port to use telnet with? Pop3 is BAD, IMAP is BAD. You should be using the encrypted versions of all of these. Anything you can't secure directly should be tunneled via port forwarding with a ssh command. Not everyone is doing this across the Internet. : ) Not that I use very frequently these days either. But it has come handy a couple of times this century where I lacked any other tool to easily verify connectivity. You cannot use ssh to check responses on most ports. But otherwise I agree with you. Unfortunately, as we all know, it's not a perfect world where we can always have our way. (If so MS, f.ex., would have been a good team player, and no hacker would break the law. Spammers, ... : ) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJjfAljK16xgETzkRAvs3AKCApBgqC+SCdQypanhU7WSF/dgi4QCfed4D h8H4ZzGZnv7B/tZHTdBUKGg= =U9tc -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound IAX2 calls has no music on hold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 19 August 2004 08:35 pm, Robert Barnes wrote: On Tue, 17 Aug 2004 10:19:17 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other types of calls. - -- Steve Does ANYONE have music on hold working across IAX2? Google does not return anything on the subject. Except I did see on the release notes for 0.7.0 Better support for MOH in IAX2 - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJjp8ljK16xgETzkRAio7AJ9An1A5a4wbwMLZGUNC2PY06aHBEwCgl9Hi cV0SL1xwq58HT9A5hSpB3W4= =Ny5C -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound IAX2 calls has no music on hold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 17 August 2004 10:19 am, Steve Szmidt wrote: Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other types of calls. This is an odd one. It does not make sense that I cannot get Music On Hold under IAX calls... - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJKpRljK16xgETzkRAr0QAJ9jzuZRnARpqXquPx4JeIE4f4vM5gCgsVXr Kl0NMV3TPEGNqDQ5HdudFhs= =qYNO -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 19 August 2004 04:34 pm, James Freire wrote: Could I use the Flash button to do conferencing then??? If so.. how? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: Thursday, August 19, 2004 4:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Nope, it does nothing... It's not an * problem either, the button just does nothing... I think they're planning on making it work in a future release, don't quote me on that... for now it just occupies space.. -Chris Well it does. It hangs up the connection, on my phone. Latest firmware. : ) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJRZjljK16xgETzkRAm2HAJ0aP6vFxdX+pzpAA5i0MlPGc29VKACgxVwZ MmsQ0846zAEz+bzIHCV1EKc= =qsSI -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue_log analysis
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 17 August 2004 06:49 pm, lenz wrote: Hello list, I have started writing a little log_queue parser that will display stats in a graphical way based on the involved queue(s) and a start/end date. You can see a sample analysis here: http://demo.xcept.it/xc-ast/XC-AST.htm Strings are in italian, but I guess the uploaded page it's easy to understand (just notice that Chiamate is calls). First it reports all calls taken by an agent, then all aborted calls, then all agents present during the given time period. Any suggestion or criticism is welcome. I am looking for queue_log files to try the software. Yours, l. In data Thu, 29 Jul 2004 15:59:39 +0200, lenz [EMAIL PROTECTED] ha scritto: Hello list, as I'm writing a little perl parser for queue_log analysis Hmm, I would love to see it but on konqueror all that shows up is Analizza Attività Code XC-AST. Same with Mozilla and Firefox. Sound like you might be using Internet Exploder code. And I see you appear to have a windows server. You can validate your html page here: http://validator.w3.org/check?uri=http%3A%2F%2Fdemo.xcept.it%2Fxc-ast%2FXC-AST.htm - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBI7mrljK16xgETzkRAtBwAKDIistud3M+K5TV9EEqmJ35VNno9QCfSk5U YSw+RXvWZwtmJ2ZFfzf6cKE= =BUSF -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
