Re: [Asterisk-Users] Mahler's Book - New Project
David Stude wrote: #2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can anyone recommend a reference book or site more suited to this task? For your project look no further than here: http://www.voip-info.org/tiki-index.php?page=Asterisk+Interop+Nortel+Norstar+MICS ___ 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] Mahler's Book - New Project
David Stude wrote: #2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can anyone recommend a reference book or site more suited to this task? Sorry that link is kind of dead. I have the pdf if anyone is interested. ___ 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] Just saw your [Asterisk] xJack Segfault in Asterisk
cmould wrote: Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey ___ 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 Haven't used an xJack for at least 2 years. How long ago was that post? Maybe you replied to the wrong post. ___ 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] Qs about FXO/FXS cards
Jim Van Meggelen wrote: Frankly, what is most interesting is the fact that your systems are trouble-free. Certainly if you were to ask if such systems could be put into production, you would probably be advised not to expect much. One last thing. I have a somewhat special PSTN connection. I subscribe to comcast digital phone which is dial-tone through the cable tv network! Every installation includes an APC unit that is plugged into the subcriber's power to maintain dial-tone in case of a power outage but I'd also venture that the subscriber's power is tapped to drive ringing as well. Isn't that convenient? Instead of the cable company paying the electric utility for driving ring voltage you the subscriber do that directly. There seem to be a lot of variables with these TDM400s. Cheers, Jim. ___ 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] Qs about FXO/FXS cards
Jim Van Meggelen wrote: If I may, I'd like to ask you some general questions about the environment these systems are running in. - How are these systems powered and grounded? Not optimally by a longshot. On the Athlon machine, my main machine, all the equipment is plugged into 2to3 prong adapters. A ground tester shows that there is ground but I don't think the outlet is wired for ground - it's probably "grounded by accident". The situation on the Via C3 machine is even worse. I have no ground on that outlet and no way of wiring it for ground short (no pun) of improvising some sort of ground by drilling into concrete or some such thing. I've never looked into it - just crossed my fingers and wished for the best. This system is my lower use system. I don't get many calls on it but I've never had to reboot it because of problems with the TDM400 or Asterisk. I dial out on the one phone connected to the system occassionally and I usually have availability and when I don't it's usually because the network has gone out on the Athlon box - the two boxes are iax/ethernet connected. - Are the lines feeding the FXO cards coming from the PSTN, or are they being fed by a PBX or similar? (basically, how long is the loop between the card and whatever is feeding it?) Yes, the lines come from the pstn. I pulled a run of Cat 5 along the outside of the house underneath the overhang of the cedar shingle siding for about 20-25 feet from the demarc through a drill hole to the Athlon box. You are successfully running systems that many would tell you to expect problems with. The TDM400 FXO modules are generally agreed to be an improvement over the X100P, so if you are having no troubles now, it is entirely plausible that migrating to TDM400-based FXOs will work for you as well. Unfortunately, there is no way of guaranteeing that, and it's your money, so I can't advise you much more than that. Frankly, what is most interesting is the fact that your systems are trouble-free. Certainly if you were to ask if such systems could be put into production, you would probably be advised not to expect much. You know what? To top it all off, I also use both systems as routers. On the Athlon box, I have three ethernet adapters one of which is a via-rhine embedded adapter. I run openvpn, [EMAIL PROTECTED] and a netfilter firewall along with Asterisk on that Athlon box. One adapter is connected to the cable modem and another to a WAP. The Via C3 system has two ethernet cards as well and runs [EMAIL PROTECTED] with Asterisk. Both systems are connected with a cheapie powerline/homeplug bridge. My only problem with all this is with the network on the Athlon box going down once in a great while. An ifdown/ifup is usually what it takes to fix it. There seem to be a lot of variables with these TDM400s. Cheers, Jim. ___ 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] Qs about FXO/FXS cards
This has been an interesting discussion. I'll chime in with my experience here. I have two servers. One with the cheapest motherboard and athlon processor I could find on Newegg.com. The other is a 1999 era motherboard with a Via C3 processor, again a bargain basement special. The Athlon system has a decent power supply - 400+ watt, the Via has a very generic PS that came with the case - 300 watt tops. On both system I have TDM cards, the Athlon has a 4 port FXS and two x100p's, the Via has a 2 port FXS. Both systems are in "production" if you could call it that because they handle little traffic - home/hobby systems. I have had no problems at all with the tdm cards or Asterisk. I occassionally lose my network on the Athlon machine - I chalk that up to the fact that I'm currently sharing an IRQ with two ethernet cards and an X100P. I'm thinking of ditching the two x100p's in my Athlon machine for a TDM card with FXO modules to free up a slot and hopefully the burdened IRQ. Based on what I'm reading here I probably should think *really hard* about that. ___ 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] send Flash via FXO
Ryan Courtnage wrote: Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really telco 3-way). The FXOs on TalkSwitch devices do support this feature. Small businesses enjoy it, because it allows incoming calls to transfered to home/cell without tying up 2 lines. I haven't seen options for zapata.conf that suggest this is supported on fxo interfaces. If it's not supported, is this something that could be achieved via changes to the zaptel drivers (without re-engineering the card/modules)? Thanks I believe there's an application called flash that does what you need. You'd announce on an IVR "Press 1 to transfer to my cell phone" and then execute flash on that fxo channel. It looks like the command is best used in an AGI script. ___ 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 newbie questions
John Stegenga wrote: [sarcasm on] Thank you ALL for your warm welcome to this list. I posted this message yesterday, and since I'm only getting Digest I figured I'd see a response in a day... [sarcasm off] C'mon. This is the Asterisk Users mail list, isn't it? This is where the Voip WIKI tells me to go for information on how people are using *. Even if you only point me in the direction of some other information, it would be great if I could hear SOMETHING from you guys and gals out thereI humbly seek YOUR wisdom. Sarcasm will not get you far on this list - no sir. Reposted message: Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. People (marketing folks) have pointed me at Cisco, but I really don't wanna. I'd rather be the hero and pull this off with a much smaller budget. Here is a scenario - 40 person call center, all with PC's (windows) and soft-phone. -any recommendations on hardware to run *? soft phones? 90% of calls would be IP / IAX coming to the center. See the wiki: http://www.voip-info.org/wiki-Asterisk+Hardware I read in the list archives about an ACD application / extension to * that would probably to what I need in that regard. - thoughts? I've read in the mailing list, the wiki and other websites that the ACD system is quite capable for the size of call center you have in mind. The only thing I don't know for sure is the support for soft phones. In remote locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. In a typical scenario there would NOT be a lot of simultaneous calls unless the system we're supporting went down hard. If you use a t1 card on the the PBX and a t1 wildcard on the * box one piece of cat 5 wired crossover should do the job. Spend the extra money for the T1 card on the PBX - there's a huge gray market for PBX T1 cards - ebay is a great place to find them. How would / could? one configure * at the remote location to communicate with * at the call center? Most reliable way and most expensive is to set up a point-to-point T1. Just tell * at each site where the channels are and route calls to them. Next step down would be IP ports at both sites on the same Tier 1 carrier network. Asterisk has it's own incredibly efficient VOIP protocol called IAX which has a trunking mode and would work well with this. Anything less (cable modem or DSL) you might risk reduced call quality or dropped calls but I've heard people vouch for the quality of IAX on dialup connections in third world countries How would / could? one configure * at the remote location to use the existing TDM PBX as failover to call the support center via 1-800 if the IP circuit died? Not quite sure of the scenario but pretty easily I think just a matter of routing calls to the legacy PBX upon failure or timeout in the * dialplan. I know you're all banging your heads on your desks saying "OY! another newbie". Thanks in advance for your wisdom and guidance. John ___ 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] Zaptel 'Under the Hood' Project
Greetings All, I have a new post on the blog. It goes a little bit more in depth on wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun. Take a look: http://zapteldoc.blogspot.com Regards, Victor ___ 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 'Under the Hood' Project
Hello All, I've got a new post up on the blog. I would have had it up sooner but blogspot was having trouble. I even included neat graphic! Check it out: http://zapteldoc.blogspot.com By the way someone mentioned that my choice of color scheme wasn't optimal so to speak - i.e. lacked contrast. I'm not a web designer so I was inclined to go with one of blogspot's limited theme choices. If anyone has a preference of blogspot theme or can give a succint set of instructions on manipulating the current theme - please let me know. I'll defer to your judgement. Regards, Victor ___ 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 'Under the Hood' Project
Holger Schurig wrote: I'd thought I'd been through the whole Zapata Telephony Site. Could you e-mail back and point to the specific links you had in mind? Start with http://www.zapatatelephony.org/philos.html and dive into http://www.zapatatelephony.org/project.html and then into http://www.zapatatelephony.org/conf.html ___ 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 The first two pages I've seen but I must admit the last is new to me. It must have been "hiding in plain sight". Thanks! ___ 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 'Under the Hood' Project
Hello Everyone, The blog for the project is up and has a couple of posts. Haloscan commenting is enabled. I've included a site feed but I'm a little unsure about it. See http://zapteldoc.blogspot.com. Regards, Victor ___ 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 'Under the Hood' Project
Tony, I'd thought I'd been through the whole Zapata Telephony Site. Could you e-mail back and point to the specific links you had in mind? As I recall, the tormenta driver source and a brief discussion on the linux port had the most releveant information. Thanks, Victor ___ 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 'Under the Hood' Project
Thanks for the replies. Today I'll set up a blog. It's quick and it will serve as a diary of my progress. I toyed a little bit yesterday with a docbook editor called conglomerate. It's a little rough but it may do the job. Lastly I'll be learning how the asterisk wiki works. This may be the best place for community involvement with the document. Everyone keep chiming in. Thanks, Victor Jon Bebeau wrote: Victor... You Go Boy!!! I think many of us, me at least, would welcome some doc on the underpinnings of Zap and friends. I'll be happy to be a "second set of eyes" to help edit such a document. Jon ___ 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] Zaptel 'Under the Hood' Project
Hello, After poking and prodding at Asterisk and Zaptel for over a couple years now, I've dedicated some time to actually reading the code and trying to figure it out. It's been fascinating. With the driver source on one part of the screen and a pdf of "Linux Device Drivers" on another part I've aquainted myself with device driver programming and the interesting hardware on the wildcards. I've always thought Asterisk and Zaptel were two of the coolest FOSS projects around and now that I've spelunked through the code a little bit I'm curious: Has anyone ever wrote a zaptel "under the hood" type of document, discussing how the pseudo tdm bus works, the zaptel hardware, etc? If so, please point me there. If not, I'd like to take a stab at compiling a paper or article about zaptel for a general audience, technically inclined but not hard core technical, i.e. people like me who have used asterisk but always wondered how it worked down to the hardware, spans, channels, chunks, samples level. Some help from the community of course would be great, perhaps through using a blog or wiki. Once the zaptel "dragon" is dispatched, I'd then focus on Asterisk. What do you all think? Regards, Victor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Sean, Yes, that IRQ assignment seemed strange to me too. I don't understand why the kernel wanted to assign IRQS this way. I guess it's something to do with this APIC technology. Can anyone fill me in here? By the way, thanks to everyone who has contributed to this thread. It's really helped a lot. Victor CPU0 0: 102777IO-APIC-edge timer 1:471IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 9159IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17:1995769 IO-APIC-level wcfxo, wcfxo 18: 341396 IO-APIC-level wcfxs 19: 0 IO-APIC-level EMU10K1 20: 3390 IO-APIC-level eth1 21: 8652 IO-APIC-level eth0 22:788 IO-APIC-level eth2 NMI: 0 LOC: 102728 ERR: 0 MIS: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Andrew: I tried the asterisk -vvvc suggestion and I didn't get any messages when the card died. Here's /proc/interrupts before I take out the sound card: CPU0 0: 102777IO-APIC-edge timer 1:471IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 9159IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17:1995769 IO-APIC-level wcfxo, wcfxo 18: 341396 IO-APIC-level wcfxs 19: 0 IO-APIC-level EMU10K1 20: 3390 IO-APIC-level eth1 21: 8652 IO-APIC-level eth0 22:788 IO-APIC-level eth2 NMI: 0 LOC: 102728 ERR: 0 MIS: 0 and after: CPU0 0: 14903IO-APIC-edge timer 1: 2IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 7469IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17: 111534 IO-APIC-level wcfxo 18: 111626 IO-APIC-level wcfxo 19: 104013 IO-APIC-level wcfxs 20:680 IO-APIC-level eth1 21:509 IO-APIC-level eth0 22: 41 IO-APIC-level eth2 NMI: 0 LOC: 14855 ERR: 0 MIS: 0 About load: almost impossible to tell. I was sshed into the server and running top - top was showing the system 100% idle. Then I hit a download link and bang, the card died. This is all pretty academic at this point - I think Tilghman found the problem for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Tilghman, I have a feeling we're getting somewhere. I ordered three cards the very day they went on sale through the digium website. Yes, it's revision C. I guess I'll talk to digium about this. Thanks, Victor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Steve, I have the tdm card on it's own IRQ. That's one of the first things I tried. Both of my fxo cards are on the same IRQ and they seem to hold together. It's interesting that you bring up the timing issue. Why would the tdm card be so sensitive? I can understand a drop in voice quality but dying? Another thought. Downloads are usually big tcp packets? Maybe 1500 bytes a packet? Processing them probably takes more time. I've run 300kbit streaming video through the server which I believe are smaller packets and the tdm card seems to hold up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Hello again, Thanks for the timely responses. Andrew: Asterisk doesn't dump any messages except when a call comes in and asterisk tries to ring an extension - it leaves a "device busy" type of message. I checked /proc/interrupts. The fxs card is still there after it dies, but the interrupts counter does not change over time. When the fxs card is working it is usually constantly firing interrupts. I'll check load and report back. Thanks for the suggestion about the sound card. I really don't need it in the server. I'll take it out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM Card loses Dialtone and Battery
Hello all, I've posted on this problem before. Well here goes again. I have an intel mobo with a p4 2.4ghz proc, 1GB Ram. It has built-in ethernet and vga and 6 pci slots. I dreamed of making this my household communications server: internet router, firewall, vpn and asterisk. Everything works except the TDM fxs card. Well it works for a little while and it dies: no dialtone, no ring tone. All 6 slots are filled: two more Ethernet cards, two digium fxo cards, an sb live card and the tdm card. Everything that I don't use on the motherboard is turned off: serial and parallel ports, serial ata and motherboard sound. I've got all this stuff packed in a case with a 430 watt power supply. Interesting observation #1: When the tdm card dies, the fxo cards and asterisk still carry on. People can call and can leave messages, etc. I just can't hear the phone ring and I can't use the phone either. Interesting observation #2: I think I know how to make the tdm card die. I have a pc behind one of the Ethernet cards on the server. When I do a download off the net, the tdm card dies. Keep in mind when I'm doing a download two Ethernet interfaces are working, the one to which the pc is connected and the one connected to my cable modem. I've just tried another download - I'm almost 100 percent sure I can make the card die this way. Anyone been down this path before? I'd hate to buy a linksys box just to make the tdm card happy.
[Asterisk-Users] * troubles
Hello all, Been a while since I've strolled this way. Apologies in advance if this is a common line of questioning. I've just bought a new Intel 865G based board with a P4 Hyperthreading processor. I believe I've gotten SMP set up correctly: in the menuconfig I specified SMP and told acpi to enumerate processors. Did I leave out anything? Anyway, the dmesg looks good and the server doesn't freeze or blow up. In the zaptel makefile, I uncommented the flag for SMP. My problem is that on my TDM20B, I lose dialtone after a while. One time I lost dialtone but had battery, another time lost both dt and battery. A reboot brings it back. Suggestions? TIA, Victor
