Re: [asterisk-users] Asterisk 1.4 vs 1.6
On Tue, Feb 19, 2008 at 01:49:58PM +0100, Peter Nabbefeld wrote: Alex Balashov [EMAIL PROTECTED] schrieb im Newsbeitrag news:[EMAIL PROTECTED] Seysan wrote: Hello, What is the Differences between Asterisk 1.4 and Asterisk 1.6 ? Also I mean what has made it to be in a separate Branch ? There are release notes that speak to this. Where? And is there a release history anywhere? Read the UPGRADE.txt file in the source distribution of Asterisk. -- Ulexus [EMAIL PROTECTED] pgpX6qBT52NKf.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PXE-bootable diskless Asterix distro?
On Sun, Dec 23, 2007 at 04:41:42PM +0200, Tzafrir Cohen wrote: Run entirely from RAM? So if your image is 40MB, you waste 40MB of RAM just to store it? This is where (A) becomes suddenly a lot nicer. Especially if you can use union-mouting and thus to have to use such a specific system. 40MB is a paltry amount of RAM for any even reasonably modern machine. Compare, too, the respective access times between flash and RAM, not to mention the write session limits of flash (though again, to consider this is to make another mountain out of a mole hill). The only significant downside of using a net-boot setup (e.g. PXE) for something like this is that you are then reliant on another machine for the operation of that one. Considering that you'd probably be relying on it (and the switch) for normal operations anyway, I would not bother worrying about it. -- Ulexus [EMAIL PROTECTED] pgpwb6J2T8aYm.pgp Description: PGP signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer on Snom 190
Tracy R Reed wrote: On Tue, Dec 07, 2004 at 12:02:08PM +0100, Thorben G. Jensen spake thusly: I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. I have a Snom 220 with a non-working transfer button. Not sure what the problem is. Also need to figure out how to do blind and attended transfers with it. The transfer soft-key only seems to do blind. Make sure you have Break key disabled in the web configuration. In the very most recent firmwares, I believe this is finally done by default. ___ 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 with Nortel BCM
David Hajek wrote: Thanks for your answer. We don't have to use Nortel's BCM, it is one of the option we're considering (not sure if it is still in the game now). I will ask this way, what commerical fullvoip PBX you will recommend? Unfortunatelly I can't use asterisk for this central point, but I can (and will) use asterisk on satellites offices. Can you please give some hints what vendors/makers I should not forget? 3com looks promising Thanks. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Friday, October 29, 2004 4:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk with Nortel BCM Use H.323 and in the BCM set the protocol to Other. Do you HAVE to use the BCM? It's a really horrible system. I worked for many years in tech support, and I've been involved in BCMs since the beta trials of version 1.0, four years ago. I know BCM, and I can tell you that it is one of the worst telephone systems ever produced. Check out the spec sheet: - The operating system is Windows NT 4.0 -- no really, an EIGHT YEAR OLD OPERATING SYSTEM. - The MSC card is a Norstar KSU that they put on a PCI card. That's FIFTEEN YEAR OLD technology - and it shows. - The platform is an Intel Pentium III 700Mhz, with 256megs of RAM, and a 20meg hard drive. How much do they want you to pay for it? - Many of the critical scripts in the system are DOS batch files (I am NOT kidding!). The BCM is famous for it's instability (go figure), and mind-numbingly stupid interface. Unless you have a lot of money to waste on obsolescence, I'd remove the BCM completely from the equation. If you have to go Nortel, go with a Succession (even a Norstar would be a more stable choice, and you can tie it into a VoIP gateway with PRI trunks). You might want to consider not using Nortel's VoIP technology at all -- I don't think they fully understand VoIP yet. Better would be to tie any Nortel gear into your VoIP network using legacy trunking through, say, an Asterisk gateway, like this: [NT PBX/KSU]---PRI---[Asterisk]=(WAN cloud)=[Asterisk] I wouldn't use the BCM as a boat anchor, but for sure it should NEVER be used as the core of a VoIP network - it's just a key system, and not a very good one at that! Good luck! [EMAIL PROTECTED] wrote: Hello, does anyone has an experience with connecting Asterisk to Nortel's BCM (http://www.nortelnetworks.com/products/0 1/eedge/bcm.html)? I would like to make this working using some voip protocol IAX, SIP, but it looks like Nortel's can't do that? My scenario is Nortel's BCM in central office and asterisk installations in satellites offices. Just a few of comments on the BCM. The programming is _extremely_ limited. We are not talking about a real PBX here. We are talking about a glorified key system. (This is from the mouth of the tech of the company who installed it for us, and I have since verified that this is definitely the case.) The license fees are ridiculous; the pay-as-you-grow method is just another way to gouge more money out of people. (Though I realize most commercial PBX systems are using this method, now.) Be _very_ careful that you look _very_ closely at which licenses are required for what service. There will be several that are daisy-chained and seemingly unrelated. This is definitely the most frustrating part of the purchase. Whatever you come up with for license costs, give yourself ample room for errors of omission. The system is based off of embedded Windows NT, and it is riddled with bugs and erros. We are currently using an Asterisk system encompassing our BCM on all sides. The PRI comes in on Asterisk, all of the VoIP and analog sets are on Asterisk, but the digital office sets are on the BCM. This is only because the investment had already been made in the BCM by the time I got Asterisk up and running. -- Ulexus ___ 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] KSS/BLF on Asterisk
Henry Devito wrote: I am buying a Snom phone this week. I will play with this feature and see what I can get going. I will share my findings. Note also that you must (right now) manually update the firmware to the latest (currently v3.55) because, for some reason the 3.4x which the phone updates to automatically does not work with the BLF feature. I have confirmed that v3.55 does work just fine, however. The only remaining issues I have with it are the hardware design problem of the receiver cradle and the horrible, lousy, idiotic design of the conferencing feature (no split feature...dead end road) and also of the myopic transfer feature (only designed with two calls in mind). -- Ulexus ___ 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] Rhino channel bank configuration with T100P
Leandro wrote: Hi, I just bought a Rhino Channel Bank to use with my just bought T100P. Rhino channel bank provide a autoT1 features that try to detect the framing and maybe the coding of the T1. Unfortunately I tried every combination of esf, d4 and b89zs and ami without success. My first guess would be that you are not using a T1 cross-over cable. Note that this is _not_ the same as an ethernet cross-over cable. I believe there is a wiring guide on the wiki, but if not, Google it. I tried also to manual configure the channel bank with the same result... nothing. Can you provide me some hint on configuring these pair of devices? Leandro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 190 - strange voice problems
Jeremy Rusnak wrote: Hi all, ...snip... We're running SIP and version 3.46 of the phone firmware. I don't know about this specific problem, but the latest firmware is 3.55. I'd try it. http://www.snom.com/download/share -- Ulexus ___ 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 Phones and asterisk
Uma S. Pandey wrote: Hi I am using Snom 190 with asterisk 1.0.0, and am observing that when I try to use asterisk voicemail or any application where I need to input more digits, then , asterisk is not able to get all the digits pressed from the phone. Can anyone suggest what might be happening? Following code I have in sip.conf file. Older firmwares had a problem with DTMF generation especially when in speakerphone mode. Additionally, even now, there seem to be some timing problems. Disregard all of these issues, though, and use out of band dtmf. It works much better. Set dtmfmode=rfc2833 [1114] ; type=friend host=dynamic ;defaultip=0.0.0.0 username=1114 secret= context=default nat=yes dtmfmode=inband This should be dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 canreinvite=no ___ 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] ZapRAS from both sides
Okay, it turns out I was being stupid. ZapRAS does not automatically answer the channel, so, the PRI_NET side below should have read: PRI_NET side -- extensions.conf: [incoming] exten = 6999,1,Answer exten = 6999,2,ZapRas(debug|64000|noauth|multilink|netmask|255.255.255.0|192.168.10.2:192.168.10.1) I have added this as a note to the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ZapRAS Ulexus wrote: I am flailing around here trying to make ZapRAS function reliably with an asterisk box on both sides. My interpretation of the documentation and the wiki is as follows: PRI_CPE side --- zapata.conf: minunused=2 minidle=1 idledial=6999 [EMAIL PROTECTED] extensions.conf: [idle] exten = 6999,1,ZapRas(debug|64000|noauth|multilink| netmask|255.255.255.0|192.168.10.1:192.168.10.2) PRI_NET side -- extensions.conf: [incoming] exten = 6999,1,ZapRas(debug|64000|noauth|multilink| netmask|255.255.255.0|192.168.10.2:192.168.10.1) This would seem to be reasonable: The cpe side dials out 6999 on the PRI, which goes to 6999 on the net side, which starts pppd on the net side. Then, cpe side sends the call to [EMAIL PROTECTED], where pppd is started on the cpe side. The only problem is, this doesn't work. The cpe side never starts pppd. I can see the net side starting it for every channel that the cpe calls, but it eventually times out, receiving no packets from the cpe side. This does not seem to be a fluke of configuration, since, if I change nothing but the side with the idledial,minused,minidle,idleext stuff, the same things happens in reverse. Even more frustratingly, if I configure both sides identically (except for the IP addresses reversed, of course), it sometimes works. Most of the time in this scenario, though, I get what appears to be a race condition (as I would expect), and the D channel goes up and down, ZapRAS spews calls everywhere, and nothing connects. However, this is the only configuration I have yet tried that at least _sometimes_ works. Can anyone shed any light on this process? Thanks, -- Ulexus [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapRAS from both sides
I am flailing around here trying to make ZapRAS function reliably with an asterisk box on both sides. My interpretation of the documentation and the wiki is as follows: PRI_CPE side --- zapata.conf: minunused=2 minidle=1 idledial=6999 [EMAIL PROTECTED] extensions.conf: [idle] exten = 6999,1,ZapRas(debug|64000|noauth|multilink| netmask|255.255.255.0|192.168.10.1:192.168.10.2) PRI_NET side -- extensions.conf: [incoming] exten = 6999,1,ZapRas(debug|64000|noauth|multilink| netmask|255.255.255.0|192.168.10.2:192.168.10.1) This would seem to be reasonable: The cpe side dials out 6999 on the PRI, which goes to 6999 on the net side, which starts pppd on the net side. Then, cpe side sends the call to [EMAIL PROTECTED], where pppd is started on the cpe side. The only problem is, this doesn't work. The cpe side never starts pppd. I can see the net side starting it for every channel that the cpe calls, but it eventually times out, receiving no packets from the cpe side. This does not seem to be a fluke of configuration, since, if I change nothing but the side with the idledial,minused,minidle,idleext stuff, the same things happens in reverse. Even more frustratingly, if I configure both sides identically (except for the IP addresses reversed, of course), it sometimes works. Most of the time in this scenario, though, I get what appears to be a race condition (as I would expect), and the D channel goes up and down, ZapRAS spews calls everywhere, and nothing connects. However, this is the only configuration I have yet tried that at least _sometimes_ works. Can anyone shed any light on this process? Thanks, -- Ulexus [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some photos from Astricon 2004
el Flynn wrote: Lenny Tropiano / asterisk.org Mailing list wrote: These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny Anyone knows if those Snom Keypad 220s are available, and where I might be able to get my hands on a few? I was talking to NETXUSA at the show, and they have them in stock. They also had them set up and working (though they hadn't tried the BLF (Busy Lamp Field) aspect of them when I checked. ___ 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] TDM400P FXO and Primus TalkBroadBand
Ryan Courtnage wrote: Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus -- DLink ATA FXS -- TDM400P FXO -- Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS -- TDM FXO link 'goes dead'. Has anyone had any measure of success doing this? Primus' service is becoming very popular in Canada, and some customers are wanting to do this. Not with Primus/Dlink, but I am having the same issue with by Vonage/Motorola. I have not really looked into it yet, though. ___ 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] GRSecurity and ALSA on a Gentoo Server
Deon Rodden wrote: I've been working with Asterisk for about 2 months now and am doing well. However I decided to switch platforms from Fedora Core 1, that my predacessor was using, to Gentoo, for obvious reasons. It just seems faster and less bloated everything I need, nothing I don't. Anyways, I've read what the Wiki had to say about it and I was only confused on one thing, putting ALSA in my USE statement. It's a 1U server with no Sound Card. I did not choose to put ALSA in my USE flags as I don't have a sound card. But will Asterisk suffer in any way? I know that Asterisk is fully capable of running on a machine with No Sound card, my Fedora servers have no sound card, but by ommitting alsa in my USE flags, will Asterisk be compiled in a way that would make it less functional? No. There is no problem installing or running it with USE=-alsa. My last question, sorry guys (and girls), is about the grsecurity in the 2.4 kernel (I chose 2.4 instead of 2.6). I set it to low for now, as it said it wouldn't cause any compatibility issues with 99% of the programs. Has anybody tried medium, or even high, with Asterisk? How secure can you get the kernel without interfering with Asterisk. Yes, I use asterisk with grsec on high. No problems. This is just more of a comment, but if anybody see's anything wrong with it I'd like to know. I don't want to use the 0.9.0 ebuild (but I emerged it just to get the dependencies taken care of) so I emerge'd the CVS program so that I can upgrade libpri/zaptel/asterisk from 0.9.0 to the latest. The The Wiki mentions something about CVS and points to: http://bugs.gentoo.org/show_bug.cgi?id=33345 but that link is dead. I figured I'd just CVS Asterisk the normal way, do the make install and it should upgrade it. I don't use the portage ebuild for Asterisk, so I don't really know. However, after a brief look at the ebuild, it looks like everything is in the right place. To be sure, though, I would 'emerge unmerge asterisk zaptel libpri' and install these fresh using the normal configure/make/make install methods from the CVS sources. Regards, Deon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] TDM card loses Dial tone
Same here. I, too have received replacement cards from Digium, and I have even tried replacing the proSLICs, all to no avail. Also to note: the same port on each (of three) cards always goes out first. On Thursday, 12 February, 2004 19:22, John Vozza wrote: Same here... Usually after several of these show up in my system log: Power alarm on module 1, resetting! Need to unload/reload module wcfxs in order to get the dial tone back. Happens several times a week, sometimes more frequently. John - NetRom Internet Services 973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Thu, 12 Feb 2004, Youness El Andaloussi wrote: I experienced similar problems too with a 4 chan tdm400. This seems to especially happen when you make configuration changes. It has nothing to do with runing X or no, it does not even have to do with redhat... I experienced the same problem on mandrake. One thing you have to be extra careful is when restarting, make sure that all the modules have entirely reloaded before expecting a dialtone with an asterisk debug console asterisk -r... many of the times I thought there was no dialtone and the asterisk process had gone cukoo, I noticed that configuration was not entirely reload. Yet, reloading many times seems to get some of the TDM400 channels hung. On the other hand, this problem does not seem to happen as extensively when no reloads are made ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 phones not working.
I am using 2.03o with a Snom200 with two lines registered without a problem: However, I am not using any authorization, whih could be where your problem is. from my sip.conf: [5117] type=friend host=dynamic context=sip-gb mailbox=5117 callerid=Name 5117 On Thursday, 22 January, 2004 16:04, Ariel Batista wrote: I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration files. I have followed the Snom FAQ step by step. But it's still not working. - \ \\_ Ariel Batista // / Red-Fone Communications, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI phone vs. IP phone
On Sunday, 18 January, 2004 02:04, Ken Alker wrote: Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to patch the ADSI phones back into * was not a cost concern. I'm looking for differences between the technologies independent of cost. Pretty much no. The ADSI specification was crippled from the start to specificly not compete with PBX offerings. It has one advantage of (very limited) programmability, but a phone like the SNOM has an open-source core. It also has the dubious value of being interchangeable with a regular analog phone, but that is about it. You will not get anything near the functionality and feature set of a SIP phone, and it has the further irritation that much of its signalling is both in-band and audible. It is too bad. If it were properly implemented, the concept behind ADSI is great. Unfortunately, Telcordia strikes again. -- Sean C. McCord ___ 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] Hardware for Asterisk
On Friday, 16 January, 2004 12:27, Steven Critchfield wrote: On Fri, 2004-01-16 at 06:47, Andrew Kohlsmith wrote: If you value your data, don't use software raid. If you value performance don't use software raid. If you value uptime/stability don't use any raid on IDE. That's pure bullshit -- I use software RAID *specifically* because I value my data. I don't want to buy two hardaware RAID controllers to have one sit on the shelf just in case the first dies... and if the second dies you're SOL because they've lasted long enough that they're no longer available. Linux software RAID is available on any Linux system and if the system blows up I can put the drives in another system and *not* worry about it not being detected. As far as performance goes, I have some bonnie++ tests that I've run that show that at least on the few systems I've tested, software RAID 1 beat out hardware RAID 1 (these systems were IDE, SCSI-2 and Ultra320, with DPT RAID controllers for SCSI on P4 and I think regular Promise IDE RAID controllers on P3) -- not a huge difference in speed but one that at least tosses your if you value performance don't use software raid argument. Perhaps on a _heavily_ loaded server you might be right, but then again I feel that you're stupid for letting a server get so loaded up that it can't handle the simple mirroring algorithms in addition to normal file servering functions without degrading performance to a noticable degree. I used to believe that HW RAID was the only way to go. With RAID5 I still feel that is true to an extent. However if you're just mirroring there is _no_ significant advantage to choosing hardware RAID over software RAID. Not on IDE, and not on SCSI. In fact, there are advantages to choosing software RAID over hardware RAID, as I've mentioned above. Have you experienced a hardware failure yet that you had to come back from? If you loose a drive, it is a high probability that you will loose the controller. So unless you have a add on card, or some motherboard with 4 IDE ports, you will corrupt the second drive of a mirror. If the second drive is corrupted, then you are only a hair above not having anything. If you don't trust that, check out the GOOD IDE raid controllers. You are only allowed to place 1 drive per port, and they only use 1 port on a IDE controller. Now here we are seeing that you must have had a really abnormal, bad experience, or you are not talking from experience at all. I have, in fact, used many software and hardware RAID configurations, and I have had a great many drive failures. For mirroring, I use software RAID because is greatly superior due precisely for the reliance on the controller of any given hardware RAID array. Although I think it is very far-fetched to set such a high relational coefficient of drive failure to controller failure, (since I have had _far_ more drives fail than controllers) the facts that hardware controllers are both expensive (compared to free software) and rare (compared to any machine's normal IDE ports) culminates in my use of software RAID. I can stick the good drive of any software-mirrored RAID array into _any_ other system (Linux OR Windows), boot up off my trusty rescue CD with software RAID and networking, and immediately recover data or functionality. Further, this presumes that the machine which housed the failed drive is otherwise in a non-functional state. If this is a false presumption, because I have RAIDed my boot partition the system boots just fine with only one working drive. Even better, when I get the new drive, I can simply install and rebuild the array while I am on-line... a feature not all hardware RAID controllers have. _My_ horror stories are those of single brick outhouse servers which all sorts of special hardware failing out in the field with an SCA drive and no SCA backplane/controller within 100 miles. Even the large NAS devices that use IDE have the IDE controller built into the sled that holds the drive and use PCI hotswap technology. I don't buy it that any truly redundant raid system is as fast in software as in hardware on a machine doing anything significant. In raid 1, you are double or more writing all data to the drives. in a read environment, it might be able to share the load out to more than 1 drive and help, but I don't expect it would be much better than a dedicated controller handling the load. Any load of a software raid solution takes processor time away from the processes it is trying to complete. So take our VoIP application, if I am spending time getting the voice recording to 2 or more drives and the software to get it there, you have significantly reduced the amount of time available to the CPU to handle the VoIP packets in a timely manner. This only gets worse as call volume goes up. If it is hardware raid, you know it will be a single write and the
Re: [Asterisk-Users] Exit the Directory Application?
On Saturday, 13 December, 2003 11:16, Tilghman Lesher wrote: ... Directory does not need an escape condition. If you fail to enter anything within the allotted time (see ResponseTimeout), you jump to the t extension. That makes for a rather ill solution for the poor fool (like me, often) who accidently enters the directory and starts pounding all of the usual escape keys because he is impatient. Okay, so I am a little restricted by temper... In a production environment, it is far better to take them as a proof-of-concept/development base and customize them to your overall setup than to use them out of the box. We use Voicemail() out of the box in multiple production environments. Yes. Unfortunately, so have I. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hybrid T1 Service (WAS: Channelbank Recomendation and GS102 question)
Ollie grabbed my notes of this on his excellent site: http://www.voip-info.org/wiki-Asterisk+setup+success+2 As far as for configuration, see that page and the snips below. from my /etc/zaptel.conf: -- ### Frame to NEGIA (Span 1) nethdlc=1-24 # ### T1 to Greensboro1 (Span 2) em=25-28 nethdlc=29-48 # ### T1 to Eatonton1 (Span 3) em=49-52 nethdlc=53-72 # ### PRI to BellSouth (Span 4) bchan=73-95 dchan=96 # ### to Local PBX fxoks=97-100 fxsks=101-102 -- Notice the nethdlc channels. Those are for my data. In this example, I have a frame-relay connection to NEGIA (http://www.negia.net), our ISP, on the first T1 span, channels 1-24. Then, on the second T1 span, the first four channels are EM voice trunks, while the remaining 20 channels are for data, bound to an HDLC device. Likewise for span 3. Span four (channels 73-96) are the PRI to BellSouth. That's pretty much it for the Asterisk side of the data config. You do have to make sure to uncomment KFLAGS+=-DCONFIG_ZAPATA_NET in the zaptel/Makefile . And make sure that you have HDLC support defined in the kernel (in the WAN network interfaces section). By the way, if you are concerned about the Asterisk box's power, this is all running (voice and data routing) on an Athlon XP2500+ with 512MB RAM... about an average desktop-range system. Hope this helps, Sean C. McCord Network Administrator NorthEast Georgia Internet Access (NEGIA.net) On Sunday, 07 December, 2003 10:56, Troy Settle wrote: -Original Message- From: Walker Haddock Sent: Thursday, December 04, 2003 7:54 PM To: [EMAIL PROTECTED] We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. This is a service that I'm interested in selling. Would you be willing to share with me (the list) exactly how you have this set up (read: your configuration files)? I've never used linux as a router, and am a bit leary of doing this and selling it as a supported service. I've got the voice stuff down I think, my primary interest is in how you accomplished the data portion. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
The directory is generated from the voicemail.conf, so I imagine you would also have to an entry for extension '#' to voicemail.conf as well. This seems like a really cheap (if effective and expedient) way of doing it. Just a note (and I really should add this to bugs.digium.com, I suppose), both the Directory and the Voicemail2 apps have very myopic view of the rest of the dial-plan or even their current context. Namely, the lack of an escape condition for the Directory and lack of most any dial-out conditions (i.e., '0' or another extension number) in Voicemail2. In a production environment, it is far better to take them as a proof-of-concept/development base and customize them to your overall setup than to use them out of the box. Luckily, this isn't too hard, since most of the important treeing is already handled with case statements. Just add the appropriate line... On Thursday, 13 November, 2003 11:34, Tilghman Lesher wrote: On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote: How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Just enter a new extension. For example, if you want # to exit the Directory application, program the # extension. exten = #,1,Goto(s,5) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP PUBLISH and SUBSCRIBE extensions?
I haven't actually seen a 100 or 105, but my understanding is that they do not have the soft keys with LEDs like the SNOM 200 and the really nice SNOM 220 that is supposed to be out next year with 30 something soft keys. Apparently, from what I've read, the Cisco extension monitoring LEDs don't work with SIP and the skinny drivers don't yet support it for asterisk. Surely, someone has had need of this feature for the attendant or secretary or something... I just can't find anything about it. ADSI phones were deliberately crippled to be without this feature (actually, the specification was) so as not to compete with commercial PBX/phone offerings. Finally, this PUBLISH method's pre-RFC draft was just released less than two months ago. On Tuesday, 09 December, 2003 23:33, Juan J. Sierralta P. wrote: On Tue, 2003-12-09 at 23:06, Ulexus wrote: After having received my brand new SNOM 200 phones and trying to get the remote extension monitoring to work, if seems that they use the SUBSCRIBE and PUBLISH SIP methods to do this. Does Snom 100/105 remote extension monitoring also ? I think that feature isnt in current * implementation, since it means patches on the whole code instead of a patch only in chann_sip.c. Anyway its a really nice feature ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP PUBLISH and SUBSCRIBE extensions?
After having received my brand new SNOM 200 phones and trying to get the remote extension monitoring to work, if seems that they use the SUBSCRIBE and PUBLISH SIP methods to do this. Further, doing a swift grep of the asterisk code, I don't see anything like this in Asterisk. Has anyone heard of anything about anyone working on this? Mark? http://www.ietf.org/internet-drafts/draft-ietf-sip-publish-01.txt Thanks, Sean C. McCord Network Engineer NorthEast Georgia Internet Access ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Live, production example
Just an example, then: I have, as my first outside production site, just concluded a very (in my opinion) interesting and educational install as described below. There are still many tweaks which need to be done, and if anyone has any suggestions for improvement, I am always First, the customer's request: Connect the three separate offices with voice and data in an inexpensive and supportable manner. Specifically, the customer requested that there be NO VOIP. Next, the hardware used: 1x Digium T400P $1500 2x Digium T100P's $1000 3x Digium TDM400P's each with 4 proSLICs $1050 6x Digium X100P's (workaround for PBX feature problem)$600 3x Athlon 2500+ systems, each with 512MB RAM and software RAID-1 80GB IDE HDD's, using Gigabyte 7VT600-L motherboards and running Gentoo Linux 1.4 existing KSU at each location: Samsung DCS (series--- one was a DCS compact, the other two were DCS) $1500 The connections used: 1 PRI from BellSouth with 100 DID numbers(Full 23 channels, which is way overkill-- not my decision) $950/mo 2 Point-to-point T1s (price varies on distance) $1300/mo 1 Frame relay T1 to Internet $650/mo PRI, frame, and one end of each of the P2P T1s comes into the T400P One T100P takes the other end of the P2P T1s at each remote site 1 TDM400P at each site goes to KSU trunk interface 2 X100Ps at each site go to KSU analog station-side interface P2P T1 dimensioning: channels 1-4 = EM trunks for voice channels 5-24 = nethdlc for data (PPP encapsulation) KSU interfacing: Incoming main line calls: 1) PRI - main T400P 2) Asterisk plays AutoAttendant stuff, then follows the following for DID Incoming outside DID calls: 1) PRI - main T400P 2) Asterisk A to either local, Asterisk B, or Asterisk C, based on DID 3) Asterisk - T400P 4) TDM400P - DISA trunk on KSU (DID not supported by KSU on analog trunks...argh) 5) KSU - Digital Samsung KSU station Outgoing outside calls: 1) KSU Station - KSU - TDM400P - Asterisk 2) Asterisk B,C - Asterisk A 3) Asterisk A - PRI Local calls: 1) station dials 9+extension 2) KSU - TDM400P - Asterisk 3) Asterisk to other Asterisk 3) Asterisk - TDM400P - KSU DISA trunk - KSU Station To check voicemail from anywhere in-system: 1) station dials 98+extension 2) KSU - TDM400P - Asterisk 3) Asterisk to other Asterisk, if necessary 4) Asterisk - VoicemailMain2(extension) To take voicemail, I had to use X100Ps connected as stations, because the KSU cannot forward on Busy/NoAnswer to an external number, and because I had to use DISA instead of DID, asterisk thinks the call is answered when the KSU picks up... before the station even rings. This wouldn't be a problem in a native environment, but to scimp on the cost of handsets, the client wanted to keep the old KSUs. 1) Incoming call creates a Busy or No Answer condition for the KSU 2) KSU forwards the call to an internal-to-the-KSU extension 3) this extension is connected to an X100P, which receives the KSU's DTMF voicemail routing digits (one of the saving graces of the Samsung DCS) and takes the call to Voicemail(extension) Data setup: The data side of things seems to be the least documented aspect of asterisk...probably because it isn't really in asterisk. It is a feature of the Digium cards and the zaptel drivers for them. Each location has a separate private subnet and a shared transport private subnet nethdlc over T100Ps with PPP encapsulation (this requires sethdlc from zaptel CVS tree, which doesn't appear to be made by default... just 'make sethdlc') sethdlc hdlc0 mode ppp ifconfig hdlc0 local transport IP pointopoint remote transport ip up route add -net private supernet netmask private supernet mask gw remote transport ip if asterisk b or c, route add default remote transport ip For the frame: sethdlc hdlc3 mode fr-ansi create 16 ifconfig hdlc3 up ifconfig pvc0 local public ip pointopoint ISP gateway up route add default gw ISP gateway Obviously, this proves the concept for a number of different features the the most excellect Digium hardware (Thanks, Mark and gang!) This system has been in production use for a little over a month now, and although I have had a few problems (one bad DIMM, TDM400P revision BellSouth fouling up the ownership of the PRI which caused a huge delay in getting the main telephone numbers ported over to the PRI), Digium has offered great support and with the source code so wonderfully available, I have been able to diagnose most of the problems with reasonable effort. If anyone would like further details, I do not mind sharing. If anyone would like to offer further suggestions, I do not mind receiving. ;) On Sunday, 02 November,
Re: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
To try and put it simply, the zaptel drivers will not compile with the -DZAPTEL_NETWORK flag (as set, not my default, in the Makefile), with any stock kernel including and after 2.4.21, which is when the new HDLC structure was imported from the development kernel tree. Therefore, it should be perfectly fine to run RedHat 9 or whatever as long as you installed (probably manually for RedHat) a stock kernel of version 2.4.20. Mind, however, that I do not have a RedHat box and that RedHat has historically made pretty extensive changes to a lot of the normal defaults to a lot of things, so the above statement may not necessarily be true. On Sunday, 02 November, 2003 11:22, Ray Burkholder wrote: All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this flag? Or do you not use up2date to get and latest kernel and source? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monitoring the asterisk and safe restart
I use daemontools ( http://cr.yp.to/daemontools.html ). Here, for instance, is my '/service/asterisk/run' script. You only need the first uncommented line and the last uncommented line. You can ignore all of the networking stuff, but I wanted to know, if anyone else happened to see this, if someone knew if zaptel's PPP device can use authenticated PPP (PAP/CHAT) in this case. I had to get a special setup from my ISP to get a frame without PAP authentication. /service/asterisk/run: #!/bin/bash # # Script to unload and reload asterisk and related networking functions # # Direct stderr to stdout so daemontools logger puts it all in log/main exec 21 # Bring down network connections so ztcfg can run ifconfig pvc0 down ifconfig hdlc0 down ifconfig hdlc1 down ifconfig hdlc2 down /etc/init.d/monmotha stop /etc/init.d/monmotha zap # Remove modules rmmod wcfxs wcfxo tor2 zaptel # Reload modules modprobe tor2 modprobe wcfxo modprobe wcfxs sleep 3 # wcfxo doesn't like things too fast, so wait 3 sec # Configure WAN cards ztcfg - sleep 3 # wcfxo doesn't like things too fast, so wait 3 sec # ## Configure Frame to ISP sethdlc hdlc0 mode fr-ansi create 16 ifconfig hdlc0 up ifconfig pvc0 local public ip pointopoint gateway ip up route add default gw gateway ip # ## Configure WAN link to remote office 1 sethdlc hdlc1 mode hdlc ifconfig hdlc1 192.168.1.1 pointopoint 192.168.1.2 up route add -net 10.90.45.0 netmask 255.255.255.0 gw 192.168.1.2 # ## Configure WAN link to remote office 2 sethdlc hdlc2 mode ppp ifconfig hdlc2 192.168.2.1 pointopoint 192.168.2.2 up route add -net 10.90.33.0 netmask 255.255.255.0 gw 192.168.2.2 # ## Configure route to the rest of internal network route add -net 10.90.0.0 netmask 255.255.0.0 gw 10.90.31.1 metric 2 # ## Start Firewall /etc/init.d/monmotha start # ## Start Asterisk exec asterisk - On Friday, 03 October, 2003 19:43, [EMAIL PROTECTED] wrote: Hi List, I am sorry that I may bring the old question to the community. My question is 1. How can we determine if asterisk is working normally or not ? what kind watchdog process do we have at this moment ? 2. In case the running asterisk is mulfucntion, is there any available way to auto restart asterisk ?? Please advise if you could. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A software FAX modem
This sounds like the fax resolution is incorrect. Basically, there are only two resolutions for faxes, normal and fine. The only difference in these two is the number of lines, or the Y dimension. With fine resolution, you simply have twice the lines. Unfortunately, I do not believe there is any header information telling which resolution the file is. The resolution _is_ communicated before sending the fax, however, as part of the initial communication negotiation. This basically means that, if it does not yet have the facility, the softfax application needs to record what resolution the fax is. On Wednesday, 22 October, 2003 10:49, Steven Critchfield wrote: Figured the group would like to hear this. I just faxed a sample document from a real fax machine to asterisk semi successfully. I'll consider it just semi successfully for now because either I haven't found a viewer that puts the image in proper aspect ratio or the storage is screwy. I'm thinking it may be the fact that image apps expect the file to be in X by X dpi not X by Y. Otherwise it was readable. Also I was able to take the resulting tiff file and create a sample call file that then sent the file back out to the real fax machine successfully. The output was nearly identical to the original with the exception of being darker. I'll attribute that to cheap fax machine with crappy scan head. Otherwise, Great job. So far this is my bug list. 1. Makefile uses a include and library directory from /home/steveu. 2. Shouldn't make install for the spandsp library put the headers and libraries in the proper locations so we don't have to make special include links? Basically if #2 is fixed, then #1 will not need those paths. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
Don't forget the equally important host stamp on the file. That allows you to write two different files at precisely the same time on a shared filesystem (e.g., NFS) with no race conditions. On Tuesday, 21 October, 2003 13:37, Andrew Kohlsmith wrote: There is a C Library function that will return a unique file name. (see man mkstemp) That's the best way to go. It is generally a bad design to encode any information in a file name. Better to simply use the file's date/time stamp to order the messages. I was speaking with tclark on IRC about this this past weekend. What is wrong with using Maildir/ type interfaces for voicemail? Maildir is a very straightforward, scalable and distributable way of storing things like email (and voicemail). Each mailbox has this format: ./ tmp/ cur/ new/ When a new voicemail is created, you mkstemp in tmp/ and create the file. Once it's done, you mv it to /new. When it's listened to or otherwise accessed, it's mv'd to cur where it stays until deletion. So to recap: create and manipulate in tmp/, move to new/ once done. When no longer new, move to cur/ and leave there. No funky locking, totally NFS safe and very fast, since each voicemail is just a file. There's no patents or any kind of software encumberances to this technique, either. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
