[asterisk-users] app_dictate problems
Has no one else experienced the problem I mentioned a few days ago with app_dictate? Or maybe no one is using that app. We're having a problem with choppy audio and failure of the accelerated playback feature which seems to be consistent on a couple of installs, failing with some SIP carriers and working fine with others. MOH and other audio playback features seem to work fine. What's different about app_dictate? -- David Josephson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_dictate playback problems
I wonder if anyone else is having these problems. We are running Asterisk 1.2.17, with an assortment of SIP users and peers. This is running on an 600 MHz P3 with CentOS 4.4, and worked properly in Asterisk 1.2.15. Nothing else running on the server except the usual support stuff like sshd, a mostly idle httpd, and no GUI. app_dictate works fine for recording, but on some calls during playback the audio jumps around, playing fragments of the file. Using the fast playback mode sometimes works, sometimes causes the jumping around to get worse. Incoming calls to the Dictate() application from different SIP carriers and different hard and soft phones give drastically different results. For instance, dialing in via an 01 Communications DID (resold by Broadvoice) at 831-713-4569 fails on playback (as described, just fragments of audio) every time. Dialing in via a Broadwing DID (resold by Vitelity) at 831-621-1913 works. Calling from a Grandstream phone fails, from a Cisco 7960 works most of the time, from a Motorola VT-1005 ATA always works. All other playback modes including MOH work fine. I have some clue, but not enough. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: LiveVOIP
Luki writes about choppy audio with LiveVOIP. We have an almost identical situation except that we were switched from the San Diego gateway to the Van Nuys gateway. Some improvement but still not usable for real customers. I have an open trouble ticket with them and no progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, at best one dropout every 10 seconds, usually one short dropout every one to three seconds. The comments from their tech support and CTO were that they were aware of the problem and it was a capacity issue that they were working on. There is a separate problem in that ringback tone (or any other audio sent without answer supervision being active, apparently) is not played to the PSTN side. This is not unique to LiveVOIP and has been discussed (with its workarounds) before. I don't mind their brusque attitude or the lack of user-level support, but we won't be able to use their service if they can't fix the dropouts. There is a lot of clatter here on the list about them not being a real provider but a lot of this is sour grapes from people reselling more expensive service. We'll see ... they don't have to be 100% facilities based to provide good service, but they do have to fix this issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: T1/DS1/ISDN PRI
My understanding of the T-1 TDM and the PSTN side is pretty solid, as it is mainly based off of Intel Corporation's T1/E1 Technology Primer (see Sources), but the CPE side is largely deduced from what I knew about the PSTN side. There may be holes or mistakes, so I would appreciate any corrections or additions that you can offer. Specifically, I would like a detail of the TDM - VoIP conversion process, similar to the basic T-1 TDM one I provided. You are severely confused, using wrong terminology, so it is very hard for us to understand what exactly you are trying to say and what you are asking. Now, be gentle. What someone is missing, is that TDM and VoIP aren't converted. TDM PRI's include signaling in the same bitstream. VoIP uses separate data paths for signaling and voice. The voice data can be the same, or different. The differences between a T-1, DS-1, and ISDN are subtle and not universally agreed upon. For a discussion of these issues see the following links: They are not subtle and they are very clear. Agreed. But not to him. T1 refers to the line coding on 2 physical pairs of wire to encode and carry a 1.544 Mbps datastream. T2 is four of those signals multiplexed onto 2 pairs of wire. T3 is 28 DS1's (7 DS2's) multiplexed onto two coaxial cables. DS1 is a logical concept that defines what to do with that signal, what the bits mean; each DS1 is made up of 24 DS0 time slots. An ISDN PRI is the definition that one uses 23 of the time slots for 23 voice channels, plus one DS0 dedicated to signaling. A DS3 is 28 DS1's (and is usually carried on a T3 physical layer, does it begin to make sense?) What's the diff between a T1 and a DS1 (http://pbxtech.info/showthread.php?t=1100) PRI setup (http://pbxtech.info/showthread.php?t=1250) Don't try to gain knowledge from web forums, you'll only become dumber, it is like learning about hosting by reading WHT. I feel dumber already after reading those posts. It's too bad. A lot of people without any telephone background try to make up stuff using pieces of the old terminology and wonder why they stay confused. They could look it up, but they don't. For instance DID's. DID has a specific meaning and inward service from the PSTN handed off on VOIP isn't it. There's no difference. DS1 is a standard signaling with 1.54Mbps raw capacity. T1 is a product name for DS1 by your carriers. PRI is primary rate ISDN which is DS1 partitioned into 23 Bearer channels for calls, and one Data channel for ISUP call signaling. Not quite. See above. T1 is the physical interface, DS1 is what you carry on it. - Is my understanding of using the same codecs and signaling protocols on both sides of the Asterisk server in order to circumvent transcoding and conversions on the server correct? Yes - Are there any other host-intensive processes that I should consider offloading to the gateway, such as echo cancellation? Yes, echo cancellation. - What does the PCM µ-law codec used in T-1 multiplexing map to in terms of Asterisk codecs (G.711 µ-law, perhaps)? Yes, PCM u-law codec is exactly the same as G.711 ulaw. And outside of North America and Japan, a-law is used. - What codec does the Monitor application use when digitally recording calls (if possible, I would like to avoid transcoding the streams when recording and let sox handle the conversions on a different box)? I *believe* that it will write the data in G.711 format. Don't rely on this though. No. It writes data to whatever format the sound card supports, usually 16 bit linear (raw) which becomes .wav if you add file headers to it. -- David Josephson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
It is my understanding that TDM is circuit switched and VoIP is packet switched. It would seem to me that at some point in a TDM-VoIP gateway, a change from circuit switching to packet switching is happening, and vice versa depending on the direction of the signals. I was just wondering if anyone could detail that process and tell me if it is resource-intensive. If I'm completely off-base, please point me in the right direction. Not off-base, but you haven't made it all the way home yet. This is another layer of the puzzle, and again we are not talking about apples and apples here. Circuit switched means that there is a (real or virtual) circuit that takes data on an input port and delivers it to an output port somewhere. Packet switched means that each packet of data is examined by each port it passes, to see where it should be sent. Normally this layer of VoIP traffic is handled not in Asterisk, but in a router. You could run the router on the same Linux box that's running Asterisk (and send packets to different Ethernet ports depending on their destination address) but normally this task is handled by a separate router. There is a small computational overhead associated with adding and decoding Ethernet packets but the main routing work is done outside Asterisk, and isn't too intensive. You could read up on TCP/IP routing and understand how this works in more detail. It's too bad. A lot of people without any telephone background try to make up stuff using pieces of the old terminology and wonder why they stay confused. They could look it up, but they don't. For instance DID's. DID has a specific meaning and inward service from the PSTN handed off on VOIP isn't it. Do you have a good, reliable source that I could take a look at? It's not something you can take a look at in my experience. Some of the Bell System training material that comes up on eBay is good. You need to follow the progress from circuit-switched voice telephony circa 1930 through modern TDM, and then look at the development of TCP/IP switching separately. Is there a way to specify the format? What if there is no sound card on the Asterisk server? No sound card, no monitor. Recording to the various file formats is possible, as Herman mentioned. Your reference picture is fine ... but note that Asterisk can be the TDM/VoIP gateway, particularly when Digium releases their DS3 card (644 voice channels!) working, a lot more cheaply than a standalone box from some hardware vendor. ___ 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] LiveVoip status report
There has been improvement in the quality of LiveVoip connections. Still some packet loss and resultant choppy audio, a little worse than with Vonage or Broadvoice. As noted in several posts over the past months, they still don't handle indication of ringing on an IAX channel if the caller has dialed a number in the Asterisk switch (for instance with the DISA app). The workaround previously suggested, to Answer() and then run Ringing() doesn't work in this case, because it still sends the IAX command for ringing which LiveVoip doesn't recognize. However, Playtones(ring) does work and represents a usable workaround for the price. They claim to be working on a new session controller that will fix this and other problems. We'll see. ___ 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] Tonelist questions
In some tonelists, as used in Playtones or indications.conf, I've seen a notation to set levels, for instance [EMAIL PROTECTED] The -10 doesn't seem to do anything. Is there a patch that will enable setting levels in a tonelist? ___ 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: Ringing problems was [Asterisk-Users] TDM400P Revision question.
Rich Adamson responded to an earlier reply (not from me) Eric, those links have nothing to do with his stated problem. The problem is 105v AC on the pstn line when on-hook and no ringing. No, he says the issue is about ringing and strange voltages on his Digium TDM400 FXS ports, not the PSTN line. He measures 107 VAC on the line with the phone on hook and not ringing. Are we sure this is measured across the line and not from one side of the line to ground? If there were 107 volts AC on the pair, and there was any current available, the phone would ring all the time. Since it doesn't, it's likely to be a faulty meter or (more likely) a high resistance ground fault in the PC power supply providing a sneak path for a few microamps of power line voltage to get to the pair. Check grounding of the PC case. The basic problem however is that (assuming the meter is right) there is only 45 volts during the ring interval. This is not enough for most ringers. Try boostringer=1 as suggested by the last response and check the ring voltage then. Many ringers need at least 75 volts to function properly. ___ 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] LiveVoip incoming, no ringback still
I see in the list archives that this problem came up before, but there was no fix for it. Any clues now? Inbound calls from LiveVoip work (I am assuming they will soon fix their packet loss issues at the San Diego pop) except for one thing -- no ringback when the called extension is ringing. My inbound context gives the caller a message and DISA dialtone, DTMF digits decode OK, the extension rings and if the call is answered, it works. If it's not answered, voicemail works. But no ringback tone to the caller. Other incoming SIP and IAX calls get ringback. If I set DIAL_OPTIONS to m, the caller gets music on hold during the ring interval, but setting it back to r yields silence. ___ 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] Limitations of aah
Dean Collins asks about limitations to aah/AMP that keep some people from adopting it entirely, and resorting to editing the config files directly. I'm one of those, but it's getting a lot better. Note, I tried 0.3 for an experiment, began again with 0.5 which works fine (with 4 clone FXOs, some different sip phones, an ATA and interface to various iax and sip peers) and am planning to migrate to the next version soon. One of the good things about aah is that unlike other precooked installs, it actually compiles the installation on your box, which verifies that you have a development environment so that you *can* make patches etc without worrying about whether you have the right compiler, libraries and utilities. And yes, you can edit all the config files directly from within AMP. I just find it usually quicker to do it from the shell. If someone is fluent with vi or nano, it's hard for them to accept a GUI cut-n-paste editor. The only thing that's really missing is the command to reload whatever is needed to implement changes that have been made. A plain reload doesn't see zap channel changes for instance. When you edit those files, it should know to restart * when needed. My biggest complaint is the lack of documentation. There will (probably, for most people) always be things we want to do in the dialplan that need to be written from scratch rather than drop-in from AMP. It would be nice to know where the globals defined in each [filename]_additional.conf were used by aah and AMP; comments on each line would be great. The AMP team should concentrate on documenting the AMP interface and let the Asterisk doc team work on documenting the commands. And FOP is another set of docs -- it took me some time to find the config files, and who knows what's possible within them? Let us know how to use AMP to pass stuff to * unless you are willing and able to track all the apps within it at once. -- David Josephson ___ 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] Multiple outgoing calls through VOIP providers
Trying to get some straight info from the VOIP providers is difficult. Say there's a small Asterisk switch and it's registered with Broadvoice or LiveVOIP or someone. There are a couple of people using the switch, one is on an outgoing call with the VOIP provider. What happens when someone else initiates another outgoing call through that provider on the same SIP registry? Does * know that the SIP account is busy or does it dial out anyway? Does the provider care? Do I establish a call group of SIP accounts like I would of Zap trunks and Dial/g1 ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Square key KTS app on *
On Fri, 2005-03-25 at 08:00 -0800, Mark W Wood wrote: I have searched both the wiki and googled looking for a solution to a square key configuration. I need to have C.O. lines to appear on the buttons to facilitate a small office. All of the users can see each other and calls are put on hold and picked up by the other users instead of transferred. Has anyone done this? Can it be accomplished and how is it accomplished? Thanks in advance. There is already a bounty posted on the voip-info wiki on this, http://www.voip-info.org/tiki-index.php?page=Multiple+line+appearances+on+a+single+line+ADSI+phone The problem is not in Asterisk -- this sort of functionality could be built easily -- but in the phone. If you have a phone with multiple line appearances, emulating a key system for line selection and outbound calling is not a problem, you just have each button programmed to call an extension when you pick it up, and map that extension to the DISA application. The problem is how to emulate the busy lamp field (BLF) so the lights on the phone read out the status of the lines. This really depends on how the phone implements ADSI or other protocols to control the lights etc. on it. Find a phone with the right addressable lights and displays and it shouldn't be too difficult to pull the information out of * to send to the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 8, Issue 216
Can I program a specific C.O. line directly to a button? Adopting the Critchfield style for a moment, no, *you* probably can't. But, depending on one's expertise with the hardware and Asterisk, it can certainly be done. I am by no means an Asterisk expert but have about 35 years of random phone experience. I have an ancient Siemens HiNet SIP phone here with buttons on it that I can push and get dialtone from any of my * trunks. Any SIP phone with programmable buttons should be able to do that (hint: an s extension to a context that only accesses one trunk, or DISA) First turn off the HTML email, please. Next spend a bit of time thinking about phones you have seen that use standard analog ports. It isn't a normal option to have more than 1 line, and unless you are getting into cheap shared signal phone setups. you usually don't see more than 2 lines. The term square key is from 1A/1A1/1A2 key set terminology, 1950's to 90's more or less. If you go the SIP route, you might get a little further down the road, but you also take a chance of integrating echo into the system. The answers are there for someone with the requisite background knowledge of the hardware available. Yes. This particular answer is dependent on the programming capabilities of the instrument in question and of the programmer in question. ___ 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] Message waiting/station busy conflict?
Greetings list, We are having a puzzle with * (asteriskathome 0.5) and SIP phones (SPA2000 ATA's). If callwaiting is enabled, everything (including call waiting) is normal. If callwaiting is turned off, the phone will not accept incoming calls and the call goes straight to whatever is programmed for the busy voicemail response. It doesn't matter whether reinvite is on or off, or whether the phones are registered or not. database show looks like /CALLTRACE/2369 : 2368 /CALLTRACE/2370 : 2368 /CW/2368 : ENABLED /CW/2369 : ENABLED /SIP/Registry/2368: 6x.xxx.xx.xx:5060:3600:2368:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/2369: 6x.xxx.xx.xx:5061:3600:2369:sip:[EMAIL PROTECTED]:5061 Here is the relevant snip from the log. What did I leave out? Mar 17 12:12:09 VERBOSE[1348]: == Manager 'admin' logged on from 127.0.0.1 Mar 17 12:12:09 DEBUG[1348]: Manager received command 'command' Mar 17 12:12:09 DEBUG[1348]: Manager received command '' Mar 17 12:12:09 DEBUG[1348]: Manager received command 'Logoff' Mar 17 12:12:09 VERBOSE[1348]: == Manager 'admin' logged off from 127.0.0.1 Mar 17 12:12:09 DEBUG[1348]: Unable to find key '2369' in family 'CW' Mar 17 12:12:09 VERBOSE[1348]: dialparties.agi: Extension 2369 has call waiting disabled Mar 17 12:12:09 DEBUG[1348]: Unable to find key '2369' in family 'CFB' Mar 17 12:12:09 VERBOSE[1348]: dialparties.agi: Max calls of 1 exceeded - deleting from dial Mar 17 12:12:09 VERBOSE[1348]: dialparties.agi: Dial string is empty - nothing to do Mar 17 12:12:09 VERBOSE[1348]: dialparties.agi: Was direct call, jumping to priority 23 Mar 17 12:12:09 VERBOSE[1348]: -- AGI Script Executing Application: (NoOp) Options: () Mar 17 12:12:09 VERBOSE[1348]: -- AGI Script dialparties.agi completed, returning 0 Mar 17 12:12:09 VERBOSE[1348]: -- Executing ESC[1;36;40mWaitESC[0;37;40m(ESC[1;35;40mSIP/2368-f359ESC[0;37;40m, ESC[1;3 5;40m2ESC[0;37;40m) in new stack Mar 17 12:12:11 VERBOSE[1348]: -- Executing ESC[1;36;40mVoiceMailESC[0;37;40m(ESC[1;35;40mSIP/2368-f359ESC[0;37;40m, ESC[1;35;40mb2369ESC[0;37;40m) in new stack Mar 17 12:12:11 DEBUG[1348]: Difference is 39520, ms is 4960 Mar 17 12:12:11 VERBOSE[1348]: -- Playing 'voicemail/default/2369/busy' (language 'en') ___ 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] Obscure * command and audio questions
A few bits and pieces that I hope someone here has answers for. Basically, I'm in the audio business and I hope to get better control of the audio performance in *. 1. I seem to recall seeing a tonelist parameter that allowed tones to be set at levels other than 0, for instance !1000/[EMAIL PROTECTED] would generate 1 kHz for half a second at -20 dBm0. But this doesn't work and I can't find the original reference. Anyone? 2. In ztmonitor there is a bargraph for setting levels, and there's a patch available in bug 2783 to show numeric values for the levels on tx and rx paths. Has anyone calibrated these numbers to dBfs or dBm0? They look like a straight decimal conversion of the audio bitstream but how literal is this? 3. Levels from app_milliwatt and Playtones are different. app_milliwatt says it's generating a mu-law bitstream but I can't find the code that's actually doing this. What does Playtones use? 4. Is it possible to tweak the gains of the transcoding modules without patching the source? What about non-transcoded calls like straight a-law to a-law connections, for instance? 5. In various docs rxgain and txgain in zapata.conf are said to be in dB or percent or some arbitrary number. My C is very rusty, but in chan_zap.c the number seems to be converted to a linear gain number using a formula for voltage gain rather than power gain. Has anyone calibrated this? Thanks David Josephson ___ 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] Sangoma and other ISA T1 cards
There is a mention that the current Sangoma T1 cards (A10[1,2,4]) work with * using their WANPIPE drivers. Has anyone used any older Sangoma cards that also support WANPIPE ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FRS over *
On the various technical issues raised here (OK, we posted the rules, we won't discuss the legality anymore) I think there is only one main obstacle to using FRS radios for extensions on *. They are simplex (push-to-talk, release-to-listen). The protocol for dealing with voice-activated-switching (VOX) has been used in ham and public safety simplex autopatches but it's really tricky to get right. It would be much easier if you used GMRS or business band radios programmed to transmit on one frequency and listen on another, that way the base can be set up with one pair of frequencies and the portables on the opposite split. Yes, you need two antennas or a duplexer, but this would be a better way to do it than trying VOX and carrier interruptions to make it work simplex. Some of the cheap FRS radios have surprisingly good RF performance, I wouldn't be too put off by hams telling you they will self-destruct. Yes, they are programmed to time out after a few seconds or a minute of talking, but you will need to make your transmissions brief anyway so you can hear what the other people on the channel are saying. They aren't as fragile as they look. FRS is also narrow band (3 kHz deviation, 11.25 kHz channel bandwidth). I don't think you would be able to get more than about 1200 bps of data over this channel. Garmin does transmit GPS data over GMRS at that rate. The modern (last 10 years or more) commercial Motorola radios don't need a special programmer, just an interface box and PC software (which can be expensive). But only a few of these have the narrow band mode needed for compatibility with FRS (and none are legal for FRS as they are not type-accepted as such). Nearly all are OK for GMRS though. These were very expensive radios though, and while you might get them on eBay for $10, the batteries will be much more than that. Only one brand of cheap GMRS radios that I've seen (Garmin) has the duplex mode that allows use with repeaters and duplex base stations. I think this is essential for successful integration with a phone system. My recommendation would be to use a duplex base station radio on a low power business band channel pair. Any of the low power UHF repeaters would be OK for this (the repeater logic is all handled by app_rpt). You can get a license for the itinerant frequencies that costs just a little more than a GMRS license, and be able to use real portable and mobile radios and real antennas. There are plenty of these available with dtmf pads so you could have full control of your * switch. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 304
Daniel Nystrom wrote It seems like the Radio discussions is closing in on something I was interested in. How about controlling 30 2-way radios via E1 and 30-channel Mux (channel bank?) with EM signalling? I think the Mux uses CAS and each channel has Audio out, PTT, Audio IN, Busy. 6-wire connection i guess? That should be a really nice setup with Asterisk! Anyone tried something like this? We didn't do it with Asterisk, but a group I have been part of for about 30 years is doing this in northern California using homemade crossbar switches and Mitel PBXs and T1. Obviously we are more than a little interested in Asterisk. One last time - go read Jim Dixon's app_rpt stuff, that looks like the place to start. Now we need someone to write a module to emulate an IMTS base station and our telco will be complete ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FRS radios on *
Rich Adamson writes GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 CFR 95.141). There has been a lot of talk from lobbyists to clarify this rule, but as it stands you could conceivably connect a *private* network to GMRS or MURS radios (you can't make any plugins or modifications to an FRS radio that isn't type accepted with the radio, so connecting a phone line or * box would be out). The language is vague, see the history at http://www.provide.net/~prsg/ Would plugging into the headphone jack with a phone-patch-type device be considered a modification for radios with vox capability? It seems dumb, but that's the way the rules are written. A patch would be other apparatus wouldn't it? Sec. 95.*194* (*FRS* Rule 4) *FRS* units. (a) You may only use an FCC certified *FRS* unit. (You can identify an FCC certified *FRS* unit by the label placed on it by the manufacturer.) (b) You must not make, or have made, any internal modification to an *FRS* unit. Any internal modification cancels the FCC certification and voids your authority to operate the unit in the *FRS*. (c) You may not attach any antenna, power amplifier, or other apparatus to an *FRS* unit that has not been FCC certified as part of that *FRS* unit. There are no exceptions to this rule and attaching any such apparatus to a *FRS* unit cancels the FCC certification and voids everyone's authority to operate the unit in the *FRS*. (d) *FRS* units are prohibited from transmitting data in store-and- forward packet operation mode. [61 FR 28768, June 6, 1996, as amended at 68 FR 9901, Mar. 3, 2003] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FRS and GMRS via *
You don't need to reinvent anything to tie radios to *. Ham systems like IRLP, Echolink, eqso etc all have fairly tight controls to keep from being abused (although with a little Linux knowledge, the IRLP package can easily be used to set up your own network using their protocol). Jim Dixon seems to have done the work to integrate radios with * already. See http://www.zapatatelephony.org/app_rpt.html -- it's all there including a PCI card that interfaces to 4 duplex radios for $500 and a single card that interfaces to one radio for $100 (plus 2 FXS ports). For the rest of the details go look at the app_rpt application. There has been a lot of speculation on this list on the legalities of doing this. For amateur use under Part 97, interconnection with the PSTN is fair game *if* a licensed amateur is always in control of the transmitter and the transmissions are strictly non-business. Non-amateurs can talk over the system but only when the transmitters are being controlled by licensed amateurs. The line in the sand is that you can order pizza over the autopatch, but you can't order cheese for your pizzeria. GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 CFR 95.141). There has been a lot of talk from lobbyists to clarify this rule, but as it stands you could conceivably connect a *private* network to GMRS or MURS radios (you can't make any plugins or modifications to an FRS radio that isn't type accepted with the radio, so connecting a phone line or * box would be out). The language is vague, see the history at http://www.provide.net/~prsg/ FRS and MURS radios may be used for business without a license. See http://www.provide.net/~prsg/frsrules.htm ... GMRS may be used for business too, but employees who aren't family members must have their own licenses. Certain commercial business radio channels in the Private Land Mobile service may be interconnected with the PSTN outside of major metropolitan areas (specifically, more than 75 miles from the population centers listed in Part 90 of the rules), or with agreement from all co-channel licensees in your area, and also on low power itinerant channels. It's not trivial to get a frequency coordination and license but you can run your own mobile telephone service under these rules and conduct your regular business that way. The radios are a little more expensive than FRS radios but you do have a bit more privacy and control over what happens. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Radio over *
Pete VK2YX writes I've been involved with IRLP for about 5 years and am one of the original install team. I've gone through the emmotions of allowing other networks connect to IRLP and I know its caused some lots of headache. As far as a closed network goes, yes there is LOTS of passion to keep it HAM only and I'd have to support that notion. As far as non-radio users go, Me too. But separate IRLP (Dave Cameron's project) from IRLP (the concept). There's nothing preventing someone from using the idea for a non-ham network, in fact there are several already (eQSO for PMR446, the European equivalent of FRS). And Jim Dixon has made a parallel effort AllStar using Asterisk. I'm sure there are more. Once again, go look at the app_rpt documentation at http://www.zapatatelephony.org/app_rpt.html and look at the app_rpt right their in your Asterisk distro. DingoTel is the same sort of thing but with a voice recognition front end and voice activated transmit/receive switching VOX. I haven't had much luck in applications where one end of the conversation is someone who isn't experienced with VOX. 73 de WA6NMF ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: list SNR
John Novack writes Dare I suggest that a MUCH better job of documenting would go a long way towards eliminating the problems you mention? Now I realize that programmers are much more interested in writing code than documentation, as well as moving on to the next hot feature than making sure the current set work well, but . . . I have found the Asterisk handbook Version 2, to be kind, poor, and the Mahler book sold for a small fortune obsolete as well as lacking. Most of what is published in that book is a rehash of what is available on line free for the taking. Hi John - same experience here. Actually there were a couple of good efforts in this direction which have now merged. There are two volumes up now and more on the way, see www.asteriskdocs.org. Join and help make it happen. I have suggested that another volume ought to be about interfacing to carriers, legacy switches and subscriber equipment, etc. ___ 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] More *@Home puzzle
Is there a configuration difference for clone X100P cards versus compatible? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel = 1 without the semicolon. Any outgoing call attempt returns all circuits are busy announcement. Edited extensions_additional.conf so the outgoing macro calls channel 1 instead of group 1, no change (except the CLI reports it's dialing Zap/1 instead of Zap/g1) Console reports that it's executing Dial/Zap/g1/2345678 where that's the number I dialed, and then everyone is busy/congested... zttool reports Generic Clone Board 1, no alarms, internally clocked, no misses or violations. ztcfg - reports Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Incoming calls are not answered. I uncommented the debug=yes line in /etc/sysconfig/zaptel, and now I can see RING! NO RING! etc when the incoming line is ringing, and NO BATTERY! BATTERY! after it stops. But no activity on the console CLI when this is happening. Probably something simple and stupid, but I can't see it. Anyone else with this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100P problems
Yes - the problem was a missing signalling line in zapata.conf. Now in and out work. Also, it was news that reload from the console doesn't reflect changes made in zaptel and zapata.conf entries. Any other config files that it doesn't reload? From: [EMAIL PROTECTED] [EMAIL PROTECTED] Did you go into AMP and configure some place for incomming calls to go? --- David Josephson [EMAIL PROTECTED] wrote: Is there a configuration difference for clone X100P cards versus compatible? I have a similar problem to what David Shaw posted earlier ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: card dialer phone
Rob at draughon.org writes I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model 2662A1M; it has five lines, a hold button (I presume), card dialer capability, and a 25-pair Amp cable for connecting to The Phone System. (The card dialer feature, IMHO, scores major geek points. If you're not familiar with it, you take a special plastic card about the size of a credit card and punch out two tiny discs for each digit in a phone number. When it's time to call that number, you insert the card in the phone, take the handset off hook, push the START button, and--voila!--the phone speed dials your party.) Jerry has already posted the basics -- this is a traditional fat wire key system phone and will work with either 1A1 or 1A2 key equipment. The first pair is tip and ring, the second pair is A-lead control, the third is for the light. When you press the hold key, the A-lead is disconnected first; when you release the hold key, the line button pops out which disconnects tip and ring. The key telephone unit (KTU) card in the key service unit (KSU) chassis detects this sequence and puts a holding bridge on the line and changes the lamp from steady to winking. There is a group of telephone collectors putting together Asterisk boxes who will be able to fill you in on all relevant details, see www.ckts.info and join the list at http://lists.ckts.info/mailman/listinfo/voip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 113
Steve Blair writes I can redirect and relay calls to numerous destinations via SER but because the Octel needs an SMDI interface for mailbox identification I am stuck, none of the solutions thus far support SMDI-SIP munging. I just started thinking about the possibility of using Asterisk with a few FXS cards to provide the gateway between SIP and the Octel. The problem is I still need an SMDI channel that is integrated with the message processing part of the gateway. Has anyone look worked or developed any working models that might help? I have spent some time looking at this. There is no SMDI support for Asterisk but someone has posted a bounty for it. You might find that you can just replace the Octel with * but that's more of a jump than some are willing to take. There is the beginning of an APE derived C++ library for SMDI. ___ 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] EM trunk card?
Has anyone found an inexpensive EM trunk card that will play with *? Looking for an interface to a legacy electromechanical PBX that's able to pass answer supervision. Docs on the X100P card would be helpful, we could probably pull EM out of that. Any ideas? ___ 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
