Re: [Asterisk-Users] IAX on windows
Hey Ron, I had you on my list to notify when there was something available. See http://iaxclient.sf.net/ Still under development, but there's working cross-platform phones there now. -SteveK On Sat, 2003-03-08 at 20:32, Ron Gage wrote: I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 - (212) 533-1775 HorizonLive.com - collaborate . interact . learn "The box said 'Requires Windows 95, NT, or better,' so I installed Linux."
Re: [Asterisk-Users] Bandwidth measurement tool: bmtools
I can't resolve this host from anywhere. Is there a mirror somewhere? Thanks, Steve Bourg On Sat, 7 Jun 2003, John Todd wrote: > > This is not specifically on-topic for Asterisk, but I have found on > many occasions while working with Asterisk that it would have been > very handy to be able to measure, with some precision, the bandwidth > being used by a particular host, port, or combination of the two. > > So, I went searching for various tools, none of which were what I > wanted. They either were too clever, or too limited in their > abilities. > > However, someone forwarded the link to this tool to me about an hour > ago, and I've been thrilled that it does _exactly_ what I want. I > can use a BPF-style filter to monitor exactly what I'd like to watch, > and it hands back results to me in "real time" down to a one-second > interval. Sometimes, a small program can make me very happy, and I > suppose after a morning full of various system problems I'm overly > happy have something that works and does just what I want it to. > > This is useful for checking to see how much bandwidth a codec > _really_ uses, or seeing what your total usage is between two IAX > hosts, or pretty much anything that requires live examination of > ethernet segment traffic. > > http://s-tech.linux-pl.com/bmtools/ > > > [EMAIL PROTECTED] bmtools-0.71]# ./rate -r 1 -f 'host 10.0.1.3 and not port ssh' > -> Currently 263.05 Bps/3.01 pps, Average: 263.05 Bps/3.01 pps > -> Currently 2706.00 Bps/17.00 pps, Average: 1486.97 Bps/10.02 pps > -> Currently 588.00 Bps/6.00 pps, Average: 1186.92 Bps/8.68 pps > -> Currently 440.00 Bps/4.00 pps, Average: 1000.00 Bps/7.51 pps > -> Currently 440.00 Bps/4.00 pps, Average: 887.91 Bps/6.81 pps > -> Currently 2080.00 Bps/16.00 pps, Average: 1086.72 Bps/8.34 pps > -> Currently 1282.00 Bps/9.00 pps, Average: 1114.64 Bps/8.43 pps > -> Currently 10385.00 Bps/20.00 pps, Average: 2274.01 Bps/9.88 pps > ^C > > > JT > > ___ > 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
[Asterisk-Users] New Asterisk System
Hello! I'm new to Asterisk, although I've had my eye on it for about a year now. I just recently installed in on RedHat 8 on a 2 GHZ system, but the sound was choppy - presumably from the onboard sound card (I read about that in the archives). So I stuck Asterisk on an old ISA system - 300 Mhz, 100 mb ram, RedHat 8, and SoundBlaster 32. While there was much improvement in the playback quality (the default greeting setup that comes with Asterisk), there were a few choppy sections - certainly not acceptable to replace our current phone system. However, as I am not (yet :) a VOIP & Asterisk expert, I would like some input from those of you who are successfully using Asterisk. 1. What would a good demo system consist of for about 5 - 10 VOIP clients? (speed, memory, sound card, etc.) 2. What would a good operational system consist of for about 120 end user analog phones coming in to a Digium card through channel banks, and about 20 to 30 VOIP clients? (speed, memory, sound card, etc.) I would like to replace our old phone system with Asterisk if it is feasible. I'm also looking for answers from people who have/are successfully using Asterisk as a PBX. Thanks for your help! Steve Lorimer kmbc.edu> IT/IS Manager KMBC http://www.kmbc.edu/ 606-693-5000 x. 109 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 cards
Hi Mark, A common problem is you aren't using CRC4 and they are, or the other way around. All ISDN E1s should use CRC4, but in some places they don't. Basically, if the Digium card doesn't work, probably nothing else will. The Digiums cards talk to most things OK. Regards, Steve Mark McKibbin wrote: I think winging it is too strong a word.so of course I have not plugged it in yet, just a hypothetical question, anyway Telstra rekon they are ETSI or at least their version of it. Regards Mark McKibbin -Original Message- From: Anthony Wood [mailto:[EMAIL PROTECTED] Sent: Friday, 13 June 2003 12:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E1 cards On Fri, Jun 13, 2003 at 01:48:13PM +1200, Peter Armstrong wrote: You need to get the ETSI or Euro version of PRA from Telstra and then it will work, they offer it as well as their quaint version of PRA access. Peter Are the E100p cards Austel approved? Or does Telstra give you written permission? Or are you winging it :-) cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red led is blinking ..
Jorge wrote: Hi, I have an E100P card. My zaptel.conf is: span=1,0,0,cas,ami,crc4 bchan=1-2 dchan=3 loadzone = us defaultzone=us after execute ztcfg red led becomes blinking, why ? Anybody known the led's table of thruth ? That looks like a pretty wacky configuration. ami? cas with a d-channel? d-channel as channel 3? That doesn't look too good. Something like: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 might be a more realistic configuration. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intercom/autoanswer, SIP, Cisco
A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP Cisco handsets has been exploited by Asterisk? I ran across an April thread that debated the availability of such feature but it died out with no absolute conclusion. Thanks, Steve Bourg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic D/41E
Hi, The D41/E does not support any sort of duplex audio path operation. That seems a major limitation with Asterisk. What functionality can it actuakky support with Asterisk? Regards, Steve Matthew John Darnell wrote: It should work, but there is a fee of $30 per channel for the software. Check the archives - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, June 14, 2003 12:44 PM Subject: [Asterisk-Users] Dialogic D/41E Hi All, OS: RedHat Linux 7.2 Machine; x86 Does Asterisk supports Dialogic D/41E? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXClient news
About 2 weeks ago, I made the first announcement to the list about the first IAXCLIENT cross-platform builds. What we had then, was a command-line application which could make a single outgoing call. We've since done quite a bit of work on the clients, and the client now has: 1) A working cross-platform GUI 2) Volume level controls, and re-tuned audio processing 3) Multiple call apparances and call progress monitoring. 4) Incoming and outgoing call support, including IAX server registration. Of course, it is still cross-platform, working equally well on Windows, MacOSX, and Linux machines. Binaries (which are updated regularly) are available from http://iaxclient.sf.net/ Source code can be found in the CVS repository of the IAXClient project page at sourceforge.net: http://sourceforge.net/cvs/?group_id=72851 You can also join the iaxclient-devel mailing list, at http://lists.sourceforge.net/mailman/listinfo/iaxclient-devel to discuss using and developing iaxclient. Happy IAXing. -SteveK -- Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 - (212) 533-1775 HorizonLive.com - collaborate . interact . learn "The box said 'Requires Windows 95, NT, or better,' so I installed Linux."
RE: [Asterisk-Users] Is it possible to do this with Asterisk?
Yes; a simple AGI script can do this. The script will have access to caller id info and can prompt them for numbers, etc. then lookup in a text file or database to find the value to return. There's a text -> speech engine also available (the name slipped my mind). As well as the usual things like say number. Steve Radich BitShop, Inc. -Original Message- From: K a z [mailto:[EMAIL PROTECTED] Sent: Thursday, June 19, 2003 9:17 PM To: [EMAIL PROTECTED] Here's what I am trying to do... First I'll have a list of 4 digit numbers like: Code:OtherCode 1234:4321 : : People will call our 800#, Have the number they are calling from read to them (ANI?) & then enter in the code (let's say 1234). If the code matches one on the list, then the OtherCode (4321 for 1234) will be read/spoken to them. With the exception of the usual recorded prompts, that's all I'm trying to do here. I would like to be able to have this system running on a 24 line card. Is it possible to do this with Asterisk? _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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
[Asterisk-Users] Problem with CID matching
I'm having a problem with Caller ID matching. The call is coming in via IAX2 to our system, the caller id doesn't seem to parse right. I just got the latest CVS version an hour ago or so. Relative extensions are pretty simple: [disaid] ; ; Check caller id for disa access ; exten => s,1,Wait,0 exten => s/7031234567,1,goto,disa|s|1 exten => s,2,congestion [main] exten => _13019876543,1,Goto,disaid|s|1 I've tried a few variations. IAX2 debug shows: VERSION : 2 CALLED NUMBER : 13019876543 CALLING NUMBER : 7031234567 ANI : 7031234567 LANGUAGE: en USERNAME: xx FORMAT : 4 CAPABILITY : 2147483519 ADSICPE : 2 The result is executed congestion on the call even though the caller id matches. This should work I would think even though the caller id is coming across IAX2 instead of a zap interface. Obviously I've changed the numbers to be not real ones, but they all match like the changed values. Thanks, Steve Radich BitShop, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Video changes
WipeOut . wrote: I thought that SIP was a voice ONLY specification and that the reasoning behind the development of SIP was to do purely voice to avoid what has happened in H.323, the H.323 protocol specification grew to try and be everything to everyone (Voice, Video and Data Sharing) and so became very complex and bloated.. I may be wrong on this.. SIP basically just gets two parties connected. Whether they then exchange audio, video or whatever is largely irrelevant. The only medium dependant thing I can think of in SIP is codec negotiation. Video doesn't bloat SIP at all. Actually H.323 isn't that bloated - it just uses ASN.1 which most people think is a PITA :-( Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be there for a few months. We'd like to go ahead and replace the voicemail system with Asterisk now, and as we're ready, drop the Merlin system. My questions: Right now, the voicemail system (and auto-attendant) are connected to the switch by 4 analog lines. Logic says that these are FXS cards in the switch, like any other extension. The switch handles an incoming call and transfers it to the auto-attendant. How would such a call be identified to be dropped in the appropriate context? When the phone switch fails to reach someone at an extension, it transfers them to the voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? Thanks, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AUSTEL Certified
Rainer Jochem wrote: We plan to have certification on the new TE410P board by the end of summer. TE410P? *getting curious* What kind of board will this be? 4 x T1/E1 ports, bus mastering. One board does T1 or E1 under software control. I don't know if you can mix E1 and T1 modes on different ports, though. We are currently looking for PRI interfaces for our asterisk-box - so perhaps we'll wait a bit more (depending on what "end of summer" means ; About April in Australia :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
It is normal. What you see depends on which version of various things are on your system. The tor2 driver spends a lot of time in the interrupt service routine (about 60% of the time on the 700MHz Athlon I use). Whether the interrupt service times shows up as system usage, or falls down a hole without being reported at all, as I said, depends on which versions of things you have on your machine. Regards, Steve Steven Critchfield wrote: No it isn't normal. I have a machine with a T400P in it and I don't even see that load continuously on my machine even with calls being routed. On Mon, 2003-07-14 at 03:08, Thomas Haeger wrote: Please can anybody help me with this, have anybody experiences with the "tor2" driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with "top") that the System takes 20 - 30 % CPU usage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 quality
Jan Rychter wrote: Does G.729 provide better voice quality than GSM? (a question for people who have tried both) It depends. The bit rate of G.729 is a lot lower, so it starts with a disadvantage. To overcome that, they made it a lot more complex and tuned to the human voice. The result is for a single person talking G.729 sounds pretty good. When there are other sounds mixed in, GSM degrades more gently. The bit rate advantage of very low bit rate codecs isn't much of an advantage in most VoIP work. A single audio RTP stream carries a huge overhead, if the latency is kept low enough for a two way conversation. 8kps vs 13.2kbps sounds like a big advantage. Add the RTP overheads and the difference looks much smaller. Actually, the GSM we use here hasn't been used in GSM networks for years. The use either EFR (enhanced full rate) or half rate. EFR is some sort of CELP based codec (I can't remember the details) running at the same bit rate as the original GSM codec - same bit rate; higher quality on a single voice; less tolerant of more complex sounds. Half rate is a 5.95kbps VSELP codec. I think the half rate codec can beat the 8kbps G.729, but it depends a lot on the implementation. Nokia phones suck on half rate. Some others sound pretty good. Codec performance is difficult to compare, as circumstances affect the results a lot. Just a little background noise can often make a big difference. Codec developers spend a fortune on trials of new designs before drawing any real conclusions about them. As a crude illustraion of the problem, look at comments people post about iLBC. They range from awful to excellent. Personally, I know I would have to use such a codec a lot before making a meaningful comment. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
Matthew John Darnell wrote: Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that seems like the way to do it. You could sell it for BIG bucks. People have done this. The results are terrible. You couldn't charge big bucks. You'd have trouble giving it away. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Poll - Would you pay $30-$50 for high qualityspeech synthesis?
ak and Speechify) needed it all in RAM at once to work well (not so OK). So, you had to allow more than 200MB of RAM per voice. This may have been improved in newer versions of Speechify, but I don' t think Naturally Speaking has changed much in that time. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vendors for phones
I'm in the process of setting up a test/demonstration system to show that VoIP is realistic and applicable for our needs. We put a 7905 and 7960 on a request for quote that went out the other day (to people like CDW & Microwarehouse). All of the vendors returned thier quotes without including the Cisco phones. So my question: where do you buy your phones? We can't buy direct from Cisco (must have 3 quotes). Thanks... ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
I asked [EMAIL PROTECTED] the other day. They wrote back: > US list retail price of BudgeTone SIP phones: > Model 101 $75/ea (available now) > Model 102 $85/ea (available now) > > US list retail price of HandyTone VoIP analog telephone adaptor: > $75/ea (available in late July 2003) > >Please contact our reseller (Ovislink/dgtimes) regarding your sample >purchase. >James @ Ovislink/dgtimes can be reached at tel: (626) 854-1805 or fax: >626.854.0835 >and [EMAIL PROTECTED] Their web site is at: www.ovislink.com On Wed, 16 Jul 2003, Marian Danisek wrote: >hello, > >i found in list archives some notes about grandstream sip voip phones. >Does anybody succesfuly tested those phones with asterisk ? Mark ? >What about the prices ? > > >regards > >Marian > >-- >SUNTEQ s. r. o. >Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic >Tel: +421-46-5430 754 # Fax: +421-46-5439 144 >http://www.sunteq.sk/ > >A mind is like a parachute... it only works when it's open. > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
Moshe Yudkowsky wrote: At 10:11 2003-07-16 -0700, Chris Albertson wrote: if you want a synthetic voice to sound natural you will have to tell the software the _intent_ of the words not just the words. You would need a markup language for that I said yes The W3C has a TTS markup language, SSML, <http://www.w3.org/TR/speech-synthesis/>. However, SSML is not a _semantic_ markup language. SSML gives directives about prosidy and pronunciation. Two interesting things about SSML (which used to be called Sable). One - there is almost no support for it amongst the commercial TTS packages. Two - even the people who wrote the SSML spec don't seem to have fully implemented it. The markup in most commercial TTS software is both proprietary and cranky. > And don't put down festival. Many (most?) of the comercial systems _are_ festival. I am not putting down Festival. However, I don't believe that many or most commercial systems are based on Festival. You are wrong. All the packages I know, except Eloquence and maybe RealSpeak, are based at some level on Festival. The ones derived from Naturally Speaking have most of the Festival directories still in place. Strange, but true. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS and PBX Integration
It sounds like you want to use the IAX to provide dialtone to the Panasonic PBX? You'd use FXS cards in the asterisk box to provide signal into a CO port on the Panasonic. On Wed, 16 Jul 2003, Iván Aponte wrote: >Hi All, > >I got a doubt about something I want to do with asterisk. I have this >office (site a) with only a Panasonic analog PBX and another office >(site b) with an Asterisk Box with an ADIT 600 . I want to interconnect >both via IAX. Is it possible to put a new asterisk box in site a >without the channel bank and put a card (FXS or FXO???) and connect it >to the pbx as a CO line ? What kind of card do I need a FXS or an FXO card? > >Regards, > >Iván Aponte > >-- >Iván Aponte >email: [EMAIL PROTECTED] >Office: +58(212)9524620 >Mobile: +58(414)2774713 > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 R2 on Asterisk
LQ (Asterisk) wrote: Dear fellows, I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina. I know that the E100P don't support it right now. Correct Does anybody developing R2 drivers? Yes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is the X100P a WinModem?
Robert Hajime Lanning wrote: Ok. And the guy who started Cleveland's first ISP (former friend of mine) used to have two dozen phone lines running into his basement. What exactly is your point? How many hobbyists/hackers/etc. would have any NEED to have T-1 hardware in their house? I would. My full T1 to the internet is being installed in two weeks. Well, there are those of us who can't afford leased-lines, which I would argue is probably the majority of the people here (leased-lines to their houses, anyhow). I'd love to have a more reliable connection than cable, but cable is what's available. I'm five miles from my CO and a T would be cost-prohibitive. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is the X100P a WinModem?
Steven Critchfield wrote: If you have to ask, you aren't part of the club. BTW, $1k is a small sum for the kind of education I have received so far. Forgive me for not having the disposable income to be in the club. :) Besides, at five miles from the CO (driving distance, may be a little more or a little less if you figure wire distance), a T is not practical anyhow. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is the X100P a WinModem?
Steven Critchfield wrote: I'm sorry, either I didn't explain myself well enough or you misunderstood. I have no connection to the telco at my home. I have a T100p and a channel bank making extensions in my home. I have a cable modem connecting me to the outside world. ohh. That's different. I still wish I could afford a T-1 from the CO all the way out here. :) -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 2003-10-22 at 07:44, Andrew Kohlsmith wrote: > Can you _please_ trim the quoted text? There's absolutely no reason to > quote the entire post you're replying to, signature lines and all... +2 > points for bottom-posting though. :-) No, -10 points for bottom-posting but not trimming. If you're not going to trim, I'd prefer you save me the hassle of scrolling and top-post. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 2003-10-21 at 17:13, John Brown (CV) wrote: > Can you provide more specific information. Saying "Its Broke Jim" > doesn't provide enough content :) True that. :) My biggest complaint was how they used to sometimes take over the server's MAC address, confusing the crap out of my switch. We only detected that because we were on an HP ProCurve that we could log into and view stats on, and the MAC address kept switching between two ports. But that is fixed in the .81 release, thankfully. However, it doesn't give me much faith in their TCP/IP stack... The switch they don't work with now is a CompUSA brand 8-port switch. I don't know the model number. I admit that it's a cheap switch, but it works with everything else in my house. With the BT phones plugged in, weird things happen. When I try to access the BT web page, the phone will give me the login page fine, but when I post the password, it freezes. As in, the phone requires a hard reset, it doesn't respond at all after 20 seconds or so. I tried to look at it in Ethereal, but everything seemed normal. I have no more data than that. I replaced the switch with a Linksys, and the phones no longer lock up now. > What version of code are you running on the GS ?? 1.0.3.81 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Context restrictions
Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be able to call locally and I need long and local users to be able to call each other, and inbound calls need to be able to go to local and long users as well. I tried reading the handbook, but even though they say that you can restrict based on context, it never shows an example of how. [local-users] exten => 8478414198,1,Dial(SIP/8478414198) exten => 8478414198,2,Hangup [long-users] exten => 8478414199,1,Dial(SIP/8478414199) exten => 8478414199,2,Hangup [local] exten => _XX,1,Dial(SIP/[EMAIL PROTECTED]) exten => _XX,2,Congestion include => local-users [long-distance] exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten => _1NXXNXX,2,Congestion include => local include => long-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P stopped working
I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I rebooted a few times too, to make sure everything had been cleared out. === [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074404064]: File chan_zap.c, Line 6986 (load_module): Ignoring rxwink WARNING[1074404064]: File chan_zap.c, Line 626 (zt_open): Unable to specify channel 1: No such device or address ERROR[1074404064]: File chan_zap.c, Line 4949 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp->channel = 0, channel = 1 ERROR[1074404064]: File chan_zap.c, Line 6730 (load_module): Unable to register channel '1' WARNING[1074404064]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074404064]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. "Te contacto por fuera de la lista." Regards, Gus - Original Message - From: "Juan J. Sierralta P." <[EMAIL PROTECTED]> To: "Asterisk Users" <[EMAIL PROTECTED]> Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all "good" softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
OK. If are talking about masses of messy details, then I agree with you. Regards, Steve CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. "Te contacto por fuera de la lista." Regards, Gus - Original Message - From: "Juan J. Sierralta P." <[EMAIL PROTECTED]> To: "Asterisk Users" <[EMAIL PROTECTED]> Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all "good" softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
On Sat, 2003-10-25 at 18:49, Ken Godee wrote: > You did do a make clean first before recompiling? Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk directories and re-checking them out. Then I decided it might be a heat issue, so I turned it off for 6 hours before trying again. Still no luck. Then I figured it might be a corrupt library somewhere, or something like that, so I formatted and re-installed RH9. I still got the exact same error messages. All I wanted was the aggressive echo cancellation... Now I have nothing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
On Sun, 2003-10-26 at 08:41, Steve Meyers wrote: > On Sat, 2003-10-25 at 18:49, Ken Godee wrote: > > You did do a make clean first before recompiling? > > Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk > directories and re-checking them out. > > Then I decided it might be a heat issue, so I turned it off for 6 hours > before trying again. Still no luck. Then I figured it might be a > corrupt library somewhere, or something like that, so I formatted and > re-installed RH9. I still got the exact same error messages. I spoke too soon. After the re-install, I forgot to add fxsks=1 to my /etc/zaptel.conf. Now it works again! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ReplayTV connecting through Asterisk box
Can someone point me to the echo cancellation settings for a pure sip setup? Thanks, Stephen Subject: Re: [Asterisk-Users] ReplayTV connecting through Asterisk box > Has anyone had any luck getting a ReplayTV DVR box to connect > through an Asterisk box? Mine seems to dial just fine, but can't > negotiate a connection. I am using: I would suggest NOT using the agressive echo cancellor. I think it buggers up modems in a big way. 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
[Asterisk-Users] passing digits for voicemail from sip gateway
I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify a mailbox using digits either, it just hangs up on me. Is this a config problem on the gateway? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone got VM2 working with MySQL?
See mbranca's patch at: http://bugs.digium.com/bug_view_page.php?bug_id=441 On Mon, 27 Oct 2003, WipeOut wrote: >I guess the subject says it all.. :) > >I am running the CVS from right now.. +- 12:25 GMT > >MySQL CDR logging is installed and working.. > >Anyone got any ideas? > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. How very inefficient. Looking for peaks in the autocorrelation function requires much less compute. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Right. Dialogic and others make a big fuss of the super detection algorithms, and quote 90+% accuracy. In the real world they are utterly useless. Call answering just doesn't fall into a sufficient redular patterm. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software FAX
Lists wrote:
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over a bunch of include files to
/usr/local/include
2) build/install spandsp
3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree.
4) move Makefile.patch from oncall to apps/ dir in asterisk
5) patch the makefile
6) edit the makefile and remove all references to steve's home dir to make
it point to my spandsp source directory.
7) rebuild/install asterisk
8) Create a dir incoming/ in /var/spool/asterisk
9) edit extensions.conf and add the following line to the incoming call
contexts I have set up.
exten => fax,1,RxFAX(/var/spool/asterisk/incoming/${CALLERIDNUM}.tif)
10) create a script that emails me the tif files every time they are
received in incoming/ and delete them.
#/bin/sh
cd /var/spool/asterisk/incoming
for X in *.tif
do
if [ -f $X ] ; then
mutt -s "FAX from $X" -a $X [EMAIL PROTECTED] <
/dev/null
rm $X
fi
done
11) Add a cronjob to run my script every 5 minutes.
*/5 * * * * /usr/sbin/mailfax
12) Test and enjoy.
To send a fax all I have to do is
1) Get the .tif file on the server somewhere
2) Put a file sample.call in the /var/spool/asterisk/outgoing/ directory and
it looks like this...
Channel: Zap/3/7989106
Application: txfax
Data: /root/fax.tiff
3) Asterisk will send it or keep trying until it send it as soon as I :wq
the file in vi.
Pretty simple, I hope this helps someone else.
Brian J. Schrock
Anistone Technologies, LLC
6926 Avery Rd.
Dublin, OH 43017
Phone: 614-798-9106
FAX: 614-798-9106
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WHat is the chance, that digium will put this application into CVS?
Zero.. until its a bit more polished, and then close to 100% :-)
Regards,
Steve
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Re: [Asterisk-Users] Answering Machine Detection
Did you mistype or something. That link is about power profiling the consumption of DSPs :-) Regards, Steve Asterisk online forums wrote: some information can be found here about algorithm and descriptions of method being used. http://citeseer.nj.nec.com/393112.html Regards, Alexander *** XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] *** Unofficial Asterisk Forums *** URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
Hi Chris, That is exactly the Dialogic implementation I was referring to that was utterly useless. It works OK when people are demoing, as they always follow a certain pattern. In real like it I've always found it a recipe for screaming angry users. Depnding on your use it can get over 90% of calls wrong. It is just so dependant on exactly how people behave. Regards, Steve Chris Ziomkowski wrote: Actually, Back in '99, Dialogic used a very simple algorithm, and it was surprisingly accurate. You simply watch and see how long the initial greeting is. If it is short (say, only a few seconds), then it is generally a live person. However, if the initial greeting lasts for much longer (say 20 seconds) then you have contacted an answering machine. That is one of the big reasons CPA on Dialogic used to give so many headaches on drop and insert applications. It would sometimes wait 10 seconds before returning answer supervision to the application and the talk path would be cut through (Had to wait to determine whether it was a human or an answering machine). In this time, if a human answered, he would sometimes hangup because he wouldn't hear any response from the remote side. Properly tuned, just watching how many seconds of energy you get in the initial greeting before silence sets in will give you 90% accuracy in determining answering machine or live person. There are always exceptions however. As a first guess though, you can assume anything less than 5-10 seconds is human, anything greater is a machine. Lots of ways to get it wrong though. Not recognizing a SIT tone and returning "answering machine" for circuit failure, not recognizing when ringing has ended and misinterpreting the "hellohello" as still being ringing cadence (Dialogic did this about 3% of the time). But in theory it should be trivial to implement in Asterisk. Might want to write a new "energy detector" algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Chris At 08:24 PM 10/29/2003 +0800, you wrote: Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. How very inefficient. Looking for peaks in the autocorrelation function requires much less compute. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Right. Dialogic and others make a big fuss of the super detection algorithms, and quote 90+% accuracy. In the real world they are utterly useless. Call answering just doesn't fall into a sufficient redular patterm. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software FAX
I am taking note of people's messages about soft fax, even if I might
appear to be ignoring them. I am getting V.27ter finished off right now,
to flesh out the facilities in the software. V.27ter is used for 4800bps
and 2400bps faxes - not critically important, but useless for lousy
lines. That's seems to be nearly functional now, so I should soon be
back to fixing things.
Most of the crashes seem to be where people have an older version of
libtiff. In each case I've followed up on they have a nice new libtiff,
but still had an old version too. Older versions seem to cause trouble.
I don't intend to find out why, since newer versions are OK.
An 8 byte TIFF file means it has been opened, and a header written. The
basic TIFF header is always 8 bytes.
Regards,
Steve
Brian West wrote:
Good for you... All I can get are 8 byte tiff files.
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over a bunch of include files to
/usr/local/include
2) build/install spandsp
3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree.
4) move Makefile.patch from oncall to apps/ dir in asterisk
5) patch the makefile
6) edit the makefile and remove all references to steve's home dir to make
it point to my spandsp source directory.
7) rebuild/install asterisk
8) Create a dir incoming/ in /var/spool/asterisk
9) edit extensions.conf and add the following line to the incoming call
contexts I have set up.
exten => fax,1,RxFAX(/var/spool/asterisk/incoming/${CALLERIDNUM}.tif)
10) create a script that emails me the tif files every time they are
received in incoming/ and delete them.
#/bin/sh
cd /var/spool/asterisk/incoming
for X in *.tif
do
if [ -f $X ] ; then
mutt -s "FAX from $X" -a $X [EMAIL PROTECTED] <
/dev/null
rm $X
fi
done
11) Add a cronjob to run my script every 5 minutes.
*/5 * * * * /usr/sbin/mailfax
12) Test and enjoy.
To send a fax all I have to do is
1) Get the .tif file on the server somewhere
2) Put a file sample.call in the /var/spool/asterisk/outgoing/ directory and
it looks like this...
Channel: Zap/3/7989106
Application: txfax
Data: /root/fax.tiff
3) Asterisk will send it or keep trying until it send it as soon as I :wq
the file in vi.
Pretty simple, I hope this helps someone else.
Brian J. Schrock
Anistone Technologies, LLC
6926 Avery Rd.
Dublin, OH 43017
Phone: 614-798-9106
FAX: 614-798-9106
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Re: [Asterisk-Users] Answering Machine Detection
Ray Burkholder wrote: Might want to write a new "energy detector" algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Where does one go to learn this terminology and the math to implement it? Apparantly not to this source. :-) A Goertzel filter finds one output bin of a DFT. Since the width of the bins in a DFT is directly related to the number of samples you include in a processed block, the width of the Goertzel is too. A Geortzel is as wide or as narrow as you want it to be. Oh, and k does not need to be an integer, unless you are trying to evaluate phase. That is a common misconception. There is a sliding window version of a Goertzel filter, but this has the same characteristics as the standard version, as it is just a trick for calculating a continuous stream of Goertzels efficiently. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Three way calling problems: 2 ea. X100P 1 ea TDM10p
I'm having a problem getting 3 way calling to work correctly using two
outside lines and one extension. The two outside lines are connected
to the X100P's and a standard model 2500 phone is connected to the
TDM10.
When I dial the first outside destination 9xxx, the call completes
correctly. When I flash the hook switch and dial the second location
9yyy. The call doesn't complete, and most of the time (but not always)
the dial tone is not broken by the digit following the '9'.
My configuration files are:
#
# zaptel.conf
#
fxsks=1-2
fxoks=3
loadzone = us
defaultzone=us
#
# End of file
#
;
; zapata.conf
;
[channels]
;
; 2 ea. X100P plugged into PSTN
;
context=from-pstn
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1
txgain=1
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=no
;callerid=asreceived
channel=1-2
;
; TDM100B Port #1 plugged into analog Phone
; This phone is allowed to dial extensions and local and long distance numbers
;
context=from-analog-phones
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1
txgain=1
immediate=no
musiconhold=default
usecallerid=yes
callerid="TDM100 Zap Phone" <103>
channel=3
;
; End of file
;
;
; extensions.conf
;
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[globals]
[aliens]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up eventually.
;
exten => _.,1,Congestion
;
; Local calls
;
[pstn-local]
;
exten => _9NXX,1,Dial,Zap/1/9www${EXTEN:1}
exten => _9NXX,2,Dial,Zap/2/9www${EXTEN:1}
exten => _9NXX,3,Congestion
;
; This is the context which all house phones see
;
[house-phones]
include => pstn-local
[from-analog-phones]
ignorepat => 9
include => house-phones
;
; End of File
;
Note: I've simplified my configuration to try and get it to a minimal state
which causes the problem for clarity. The above configuration files will
exhibit the problem on the latest CVS version checked out at
8:00pst October 30 2003.
Thanks
Steve Rodgers
San Diego, CA
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Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Hi Patrick, You are in the UK, right (at least DDI strongly suggests that)? This is the commonest signalling for a DDI line on an analogue pair. The line is behaving just like the main exchange is a telephone. It picks up the line, by applying a 600ohm loop, and dials (with pulses per second or DTMF) into your PBX. Your PBX port is behaving like it is a public exchange, with a phone attached. Electrically, Digium's FXS card should do the job you need, but others will have to tell you whether * has the software features needed to make this work (it should certainly be pretty close). Regards, Steve hkirrc.patrick wrote: as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good system board to use with TE410P?
Hi Scott, I use a Tyan 2665 (7505 based) M/B with a TE410P. That works well. This is a development workstation, so its probably not the kind of board you want for deployment. Regards, Steve Scott Stingel wrote: Hi- I'm looking for an appropriate system board to power a system with two (2) Digium TE410P cards. Since these cards require the 3.3 volt PCI, I'm considering vendors like Tyan for the motherboard. Can anyone please tell me their experiences with the Tyan i7501 series (Xeon-basd), or recommend an alternate motherboard? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ 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
[Asterisk-Users] Threeway calling leaves outside trunks bridged
I think I found another interesting 'feature' with threeway calling. If you
hang up while on a 3 way call with both parties on outside lines, Asterisk
ends up removing the conference initiator and leaving the outside trunks
bridged together. Is this a good idea? This could cause congestion problems
on small configurations with limited outgoing lines. Maybe we should add an
option to zapata.conf which forces 3 way calls to be completely dropped when
the initiator hangs up on a conference with 2 outside lines bridged. Note:
if one of the conference members is an internal extension, then this
case should not not apply.
Steve Rodgers
San Diego CA
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial("Zap/3-1", "Zap/g1/9www8531212") in new stack
-- Called g1/9www8531212
-- Zap/1-1 answered Zap/3-1
-- Attempting native bridge of Zap/3-1 and Zap/1-1
-- Starting simple switch on 'Zap/3-2'
-- Started three way call on channel 3
-- Started music on hold, class 'default', on Zap/1-1
-- Attempting native bridge of Zap/3-1 and Zap/1-1
-- Executing Dial("Zap/3-2", "Zap/g1/9www2891212") in new stack
-- Called g1/9www2891212
-- Zap/2-1 answered Zap/3-2
-- Attempting native bridge of Zap/3-2 and Zap/2-1
-- Building conference on call on Zap/3-1 and Zap/3-2
-- Stopped music on hold on Zap/1-1
-- Attempting native bridge of Zap/3-1 and Zap/1-1
-- Attempting native bridge of Zap/3-2 and Zap/2-1
== Spawn extension (house-admin, 98531212, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
-- Hungup 'Zap/1-1'
-- Attempting native bridge of Zap/1-1 and Zap/2-1 -- Why can't we hang up
everybody in this case?
-- Starting simple switch on 'Zap/3-1'
-- Hungup 'Zap/3-1'
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Re: [Asterisk-Users] PHP Manager examples
Kevin Bockman wrote: Anyone have any example scripts in PHP that connect to the manager? I started a PHP * Manager API, modeled on the Perl API, but haven't had a lot of time to work on it. I'll be happy to give you what I do have. -- JustThe.net Internet & New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Hi Patrick, From memory (I haven't lived in the UK for 11 years) the electrical characteristics are pretty much the same: -48V, 35mA loop current, 600ohm complex impedance. One key difference is an extension needs to ring, but a DDI line does not. The different cards you see used may be because the DDI port has no ringer, or it may be a marketecture issue - they can probably squeeze more money out of the customers for the DDI ports. If they get full approvals only on the DDI card, then only a pricy DDI card can be used to attach to the DDI lines. Regards, Steve hkirrc.patrick wrote: Hello Steve, You are exactly right about the DDI line and thank you for clearing up the 600 ohm loop. Can you tell me other electrical details? It's just that all the PBXes (that I know) uses different cards for DDI lines and analog extension lines and since the CO normally or at least, expected to be much further away than an extension phone, I was wondering if there's a difference in the electrical requirment. thanks again, patrick Steve Underwood wrote: Hi Patrick, You are in the UK, right (at least DDI strongly suggests that)? This is the commonest signalling for a DDI line on an analogue pair. The line is behaving just like the main exchange is a telephone. It picks up the line, by applying a 600ohm loop, and dials (with pulses per second or DTMF) into your PBX. Your PBX port is behaving like it is a public exchange, with a phone attached. Electrically, Digium's FXS card should do the job you need, but others will have to tell you whether * has the software features needed to make this work (it should certainly be pretty close). Regards, Steve hkirrc.patrick wrote: as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Threeway calling leaves outside trunks bridged
You have me convinced. It's a forwarding issue not a threeway calling issue. Also, if the outgoing lines are configured for kewlstart, as long as the called parties hang up at the same time, the conference will be torn down. Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel Performance Primitives
Hi Ernest, I tried IPP, but couldn't get much performance out of it. When I tried diassembling one or two routines to see what they looked like, there seemed at be a llo of overhead in the routines that destroyed all the benefits. Regards, Steve Ernest W. Lessenger wrote: Hey all, For those of you who are really worried about asterisk performance, I thought I might alert you to a "toy" you might play around with. The Intel Performance Primitives contain a number of optimized functions for use in digital signal processing that could help with echo cancellation, codec transformations, etc. I don't have any idea how useful this would be in Real Life (actual performance gain, license compatibility, etc), but there you go... http://www.intel.com/software/products/ipp/ipp30/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get the g.723 codec?
Andrew Gillham wrote: Steve Underwood wrote: Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve I was measuring about 32-36 Kbit/s for a G.729 call, and around 20-22Kbit/s for G.723. This is at the DSL router, so it includes all of the overhead. If you're on a dialup modem, that can make or break the call. -Andrew What you are seeing their is not the codec at work, but the huge overhead of RTP. G.723.1 uses 30ms blocks. G.729 normally uses 20ms blocks. They are both usually sent with one block in each RTP packet. If you don't mind a little more latency, put two G.729 blocks into each RTP packet (I'm not sure if Asterisk supports that, but many things do). Then you have a packet every 40ms, a lower bit rate than G.723.1 and better voice quality. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
An E1 can be a long way from the box with the right cable. However many people use the wrong cable. Using a LAN cable for an E1 often gives errors if the cable is more than just a few metres long. Although the plugs look the same, the twisted pairs should be grouped differently in an E1 cable, and it really makes a difference. If the drop cable is only a couple of metres long, a LAN cable is usually adequate. This is also true for T1s. Regards, Steve Bisker, Scott (7805) wrote: How far is your server from the telco box? I found that with extended distances, my reliabilty was significantly decreased. If you still have problems, check your RJ-48X jack for connection problems. -sb -Original Message- From: Eduardo Goncalves [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 5:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Red Alarm On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz <[EMAIL PROTECTED]> wrote: Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? I'd suggest your telco doing loopup and checking the circuit. My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multitech.
Hi All, I'm new to asterisk, can I use asterisk with a Multitech mvp 210? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What the Installation Instructions SHOULD HAVE SAID..
Hello all-- When I bought the TDM400P and the two FXO cards to prototype (small-scale) what could be done with Asterisk, I got a single sheet of paper with the cards, that explained how to insert the card, and fetch the source for asterisk, zaptel, and whatnot. But, before I could get it working, there were a few things that had to be done, that are not explained on the paper! So, to help the poor newbies who are starting from scratch, I humbly throw in the following two-cents, hoping to ease their pain. Please, feel free to correct anything that I have gotten wrong. First, the sheet I got is called "Installation Instructions for the TDM400P (1-4 port FXS PCI Interface card)". The Instructions are split into "Hardware Installation", and "Driver and Asterisk Installation". There is nothing wrong with either section. Actually, anything to add would be under a third section, "Pulling it all together". "Pulling It All Together" Interrupt Priority. On Redhat Linux (and perhaps other brands), you can issue the command: cat /proc/interrupts and it will show you how the different hardware devices are assigned to the different interrupt levels or Iterrupt Request numbers, or whatever. Everything will work better if you have each card you just inserted on its own interrupt number. However, depending on your motherboard/BIOS combination, this may be difficult or impossible to achieve. You may swiftly find that the machine that will host phone hardware and Asterisk should be dedicated to this purpose, and all unnecessary options provided by the motherboard should be turned off in the BIOS -- at least, anything that requires an interrupt slot. For MY situation, everything fancy was turned off. USB taking up interrupt slots? Turn them off. Joystick, serial ports, etc? Disable them. The fewer things contending for interrupt slots, the better. Also, most BIOS setups allow you to assign interrupt numbers to cards, based on the slot you plug them into. Keep to the top slots in the system, closest to the AGP slot. Some have asserted that XT-PIC is an antiquated way to handle the interrupts, but I do not have the kernel option skills necessary to modify this so far. This is what I get: cat /proc/interrupts CPU0 0: 18311297 XT-PIC timer 1: 39583 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 182857694 XT-PIC wcfxo 5: 183672409 XT-PIC eth0, wcfxo 8: 1 XT-PIC rtc 10: 182826511 XT-PIC wcfxs 11: 45661576 XT-PIC es1371 12: 408175 XT-PIC PS/2 Mouse 14: 1688558 XT-PIC ide0 NMI: 0 ERR: 0 I personally have tried many different arrangements, and this seems the best I can do. I'd like to get the wcfxo card on a different interrupt number than the one that eth0 is on, but the system seems adamant about keeping them together. I'm using an MSI board, fairly late model. I have seen previous mailings, that seem to indicate that it is unwise to put more than 2 or 3 cards into a single machine. So, if you have hundreds or thousands of phones, split them up into groups no larger than what maybe 2 quad-span T1 (or E1) cards can handle (4*24= 96), times 2 is 192 lines per system, right? You'll have to figure out yourself how to split things up from there. It was suggested I obtain the latest, bleeding edge version of the kernel (nptl?), but I found that version wouldn't run X11 for some reason. So, I use the extent version of RH9, with all the neat RHN patches applied. The audio doesn't work at all, and I don't know why, but we'll leave that alone for now. Configuring channels and Order dependency. I have proven one thing: The order you declare your channels in the zaptel.conf file, and the order you load your modules MATTERS. Do it in the wrong order, and you will have problems. The instructions don't mention much if anything about this little detail. With this in zaptel.conf: fxsks=1,2 fxoks=3-6 I find that I should issue the commands in this sequence: modprobe wcfxs modprobe wcfxo ztcfg -v -v -v Swapping the order of the probes for fxs & fxo will issue error messages when you run asterisk. If you run into this sort of channel allocation problem, reverse the order of your modprobes, but reboot between attempts. And, if all else fails, permute the order of the boards as they are plugged into your PCI slots. Oh, and even if you get the commands in the right sequence, you will most likely get some error or warning messages from the modprobes and/or the ztcfg... If asterisk runs OK, then these can be ignored. The existing documentation is very verbose about another possible point of confusion-- FXS hardware is signalled via FXO protocols, and vice versa for FXO hardware. Notice above that the FXS card has 4 ports, and is signalled via FXO signalling. And, the cards from Digium are
[Asterisk-Users] The Minimum Cost of Setting up an Asterisk Phone System?
oip phones on an already busy data network, or to put in a separate Voip network, will affect the prices. murf -- Steve Murphy <[EMAIL PROTECTED]> Electronic Tools Company signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Hold, park, transfer, etc-- How is it done?
Hello-- I've gotten a rudimentary system up & running, and I now have these problems/questions/comments/surprises: Problem 1: In voicemail, using "6" to go to the next message doesn't seem to work at all. Problem 2: And, I specified a group to dial as an option in an extension. I have only one of the 3 extensions plugged in, and it doesn't ring (but the console says it's ringing the first member of the group.) What gives? Am I malspecifying the group (Zap/g3) in the dial command in the extensions.conf file? Or describing it wrongly in zapata.conf? Or, do I need all members of the group plugged into the card? Big Question: How do you put a call on hold? How would you transfer the call? What is "call park", and how does that play into all this? What cool features come pretty much default with asterisk? How are they accessed? Comment on Docs: The documentation available on the Digium site is nice, at least, for what is there. It'd be nice to see it finished up... in the meantime, the (ie) "show application gotoif" seems to suffice... and the mailing list has answered a few questions, like how to record messages for playback. Pleasant Surprise: I've got a 1970's rotary dial phone (which miraculously works instead of touch tone with asterisk!), and a few cheap touch-tone phones to test all this. Thanks, murf -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] my first asterisk install
Hi all, I have just installed asterisk for the first time and I got an error #1074432736 'unable to load config modem.conf' Can anyone tell me what this means, and can anyone point me to some good reading so I can get started using my adtran total access 750 with asterisk? Thanks kindly, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
WipeOut wrote: Gavin Hamill wrote: It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? No built in high availability or clustering options making it as reliable as the harware, OS and apps.. Last time I looked it up PC systems combined hardware components average reliability was about 96% uptime(This was a while back so the percentage may not be accurate).. This is a problem for telecom's system whos uptime is usually measured in years and not a percentage of 1 year.. No flames please, I realise that there are issues involved with the PSTN lines, channel banks and some other things in a clustered senario.. Later.. 96% uptime would mean nearly 4 hours per month down. I have never experiemced anything that bad using the nastiest crappiest no-name server parts. unless you want to make a point, like some authors do. Then you say the hard disk failed and it took a week to get and install a new ones, so the downtime was 24x7 hours. In reality, if your service support doesn't stock all the important bits for quick replacement, it provides no service at all. I have typically found Linux and even SCO Openserver on x86 servers have better up time than the fully redundant machines from Stratus. Their hardware may not fall over, but their OS does. When it does it takes 1 to 2 hours to reboot. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
Ariel Batista wrote: Can anyone think of any others? Here is the short list I have! 1) Lack of graphical interface. 2) Un-freindly user interface (Command prompt only) 3) Network and Telephony person needed at site. 4) No standard SIP Phone nor IAX phone available. and the biggest one I feel is a major problem! 5) Voicemail can not be configured unless you re program it yourself. And is not based on any standards! 100% of all voicemail systems are not based on standards. There *are* no standards for voicemail. There aren't even many common practices. The nearest voice mail comes to haveing a standard is that you are expected to talk after a *beep*. I think that is primarily so those who don't understand the language of the prompts can still leave messages easily. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beginners help
Rajiv--
I've just went thru something similar myself, setting up 2 FXO's, and a
4port FXS...
>From your included config, your FXO context does not tell asterisk to
answer the phone, therefore, it doesn't. You might try something like
this for your "incoming" context in extensions.conf:
exten => s,1,Answer
exten => s,2,Zapateller,nocallerid ;; try to shake off the
;; post-FTC-DoNoCallList Exceptions
exten => s,3,Background,HowdyThere ;; A short welcome message?
exten => s,3,Dial,Zap/1
exten => s,4,Voicemail(u1) ;; And, voicemail if no answer?
exten => s,5,Hangup ;; Done
exten => s,104,Voicemail(b1) ;; Busy -- Voicemail?
exten => s,105,Hangup;; Done
Or somesuch. Haven't put the stuff in priority 4-105 in the s extension
yet myself, but theoretically it should work Good Luck!
murf
On Wed, 2003-11-05 at 22:57, [EMAIL PROTECTED] wrote:
> Hi,
>
> I recently purchased two X100Ps and a TMD40B. I have set everything
> up
> according to the instructions however I can't get asterisk to pick up
> an
> incoming call (It just keeps ringing). Also, when I pick up an
> internal
> extension I get dead air. I searched the mailing list for 3+ hours --
> What
> am I missing?
>
> Thanks in advance for your help.
>
> Here are the installion steps that I used:
>
>
> /etc/zaptel.conf added:
> fxsks=5-6
> fxoks=1-4
>
>
> /etc/asterisk/zapata.conf added:
> signalling=fxs_ks ; X100P
> group=1
> context=incoming
> channel => 5-6
>
> signalling=fxo_ks ; TDM40B
> group=2
> context=internal
> channel => 1-4
>
>
>
> /etc/asterisk/extensions.conf added:
> [incoming]
> exten => s,1,Dial,Zap/1
> ;exten => s,1,Dial,Zap/5
>
> [internal]
> exten => 34,1,Dial,Zap/1
> exten => 823,1,Dial,Zap/2
> exten => 400,1,Dial,Zap/3
> exten => 500,1,Dial,Zap/4
> exten => _9X.,1,Dial,Zap/5/${EXTEN}
>
>
> lsmod outputs:
> Module Size Used byTainted: P
> wcfxo 7424 0 (unused)
> wcfxs 15808 0 (unused)
> zaptel183072 0 [wcfxo wcfxs]
> ppp_generic15676 0 [zaptel]
> slhc4864 0 [ppp_generic]
> usb-ohci 17064 0 (unused)
> usbcore36416 0 [usb-ohci]
> amd74xx 9380 1
> nvnet 25216 2
>
>
> "ztcfg -vv **" outputs:
>
> Zaptel Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> Channel 04: FXO Kewlstart (Default) (Slaves: 04)
> Channel 05: FXS Kewlstart (Default) (Slaves: 05)
> Channel 06: FXS Kewlstart (Default) (Slaves: 06)
>
> 6 channels configured.
>
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Re: [Asterisk-Users] Red Alarm
Andrew Kohlsmith wrote: An E1 can be a long way from the box with the right cable. However many people use the wrong cable. Using a LAN cable for an E1 often gives errors if the cable is more than just a few metres long. Although the plugs look the same, the twisted pairs should be grouped differently in an E1 cable, and it really makes a difference. If the drop cable is only a couple of metres long, a LAN cable is usually adequate. This is also true for T1s. Actually that's not entirely true. standard 568A/B wired cable does not split pairs for ethernet or DSX1 wiring. I've no idea what you mean here, since your next statements shows just *how* they are split. :-\ The problem is that DSX1 uses pins (1,2),(4,5) and ethernet (1,2), (3,6) (parenthesis show pairing). DSX1 must have the (1,2) and (4,5) pairs swapped to match the TX to the RX at each end, whereas normal Not usually these days. The box on the wall normally needs a striaght through cable to the card for E1s and T1s. That is why so many people plug in a LAN cable and find it almost works. ethernet does not, as the switch is cross-wired. Using an ethernet crossover cable does not help since it is swapping (1,2) and (3,6), not (1,2) and (4,5). Well, at least a crossover cable doesn't fool people into thinking they got it right. :-) The problem with using CAT5 for long telco runs is that the impedance is wrong at the line clock rate (~1MHz). IIRC the impedance for telco is specified at 600 ohms @ 1MHz, whereas for CAT5 the impedance is actually T1s are always 100-110ohm, E1s are the same when on pairs, and 75ohm on coax. Only analogue pairs are terminated at 600ohm, and no line can actually be greater than 120*PI (about 377) ohms - that is the impedance of free space. Fudgy 600ohm stuff works at audio frequencies, but you have to treat the line properly as a transmission line as the frequency rises. specified at around 100MHz, where the ethernet line rate is. You can get away with it so long as the impedance is right, but unless you've got the data sheets you're playing guessing games. There is no guessing involved. The impedances are pairing are all standard. You need specs, not data speets. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Stephen R. Besch wrote: 5) Attempt to balance the hybrid at the 2-line to 4 line interface. Why: 99% of the time, this is where the echo originates and this is where is should be fixed. Unfortunately, this is not for the faint of heart, but if your line card has a hybrid balance adjustment (many don't), use it. Also, with multiple simultaneous calls, this may be the only real solution. Part of the problem arises from the use of lower impedance telephone wiring nowdays. The typical characteristic impedance of Cat5 twisted pair is about 100 ohms and many line cards are optimized for a 600 ohm line. This is made worse if the DC resistance of the wiring to the CO switch is relatively low. I haven't tried this myself, but you might try something as simple as a 500 ohm variable resistor in series with the ring line and adjust for minimum echo. If it gets worse, you haven't lost anything, just take the resistor out of the line. If it works, measure the value of the resistor when set for minimum echo and replace it with a fixed value resistor. Tweaking the hybrid is really a waste of time. Most don't permit tweaking for this reason. Any change to the circuit, like changing to another phone (perhaps even of the same model) generally defeats the effect of any tweaking on the short lines of most PBXs. A well designed hybrid is fairly relaxed about termination, though the return loss can vary a lot across the audio band. Most approvals specs only call for about 12dB of return loss, and you will seldom see more than 20dB - even with hand tweaking. Whatever you do with the hybrid, only proper echo cancellation will clean things up well enough for good VoIP (or cellphone calls, which suffer similar high latency). Actually twisted pair are generally below 150ohms. No line can have an impedance higher than 377ohms. 600ohm termiation is a fudge. On long lines the fudge doesn't work well, and loading coils are needed to refudge things. ADSL can't work with the loading coils in place, so they are being stripped out in many places. Such is life - messy! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing an outside number -- QUESTION --
Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX numbers to go thru after the 9, unmolested. 4. After the 9, I'd like all other 7 digit numbers to be dialed 1307NXX. Simple enough, right? I wrote up the following: [outwork] ignorepat => 9 exten => _91XX,1,StripMSD,1 exten => _1XX,2,Dial,Zap/2/BYEXTENSION exten => _9754,1,StripMSD,1 exten => _754,2,Dial,Zap/2/BYEXTENSION exten =>_9NXX,1,StripMSD,1 exten =>_NXX,2,Prefix,1307 exten => _1307NXX,3,Dial,Zap/2/BYEXTENSION [workext] ;; the active context for an internal extension include => outwork ... (other extensions, but not 9) exten => t,1,Congestion exten => i,1,Congestion exten => o,1,Congestion And it doesn't work, and neither do maybe a dozen variants. I split the 3 groups up into different contexts, I remove the second two StripMSD commands. I sequence the priorities from 1 to 7. I tried setting all the post-StripMSD commands to priority 2, I... and on and on, including combinations of the above, and the best I can do, is get the first one to work. I can dial 91<10 digits>, and the <10 digits> are sent out to the phone company, and the phone rings out there. All the rest might have me wait some seconds, and then I get the congestion tone. And, on top of everything else, the moment I touch 9 on the extension, the dial tone goes away. So IgnorePat doesn't work in this situation, either. Checked the documentation for dial and stripmsd, and none of the examples cover anything more complicated than a single pattern. What am I missing? murf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Stephen R. Besch wrote: God, I wish I could be so sure of that. I've looked at the circuitry on some high end channel bank active hybrids. SPICE modeling predicts a maximum 26 dB attenuation of the returned echo, even with a balanced line. A simple circuitry change which adds a balance adjustment should permit cancellation to approach the parctical limit of a good quality op-amp (60-100 dB). I don't know what the Digium stuff looks like, so I can't comment on it. The spice modeling is rather idealised if you get those figures. Few hybrids ever give more than 20dB. If they achieve more than 12dB across most of the audio band they pass the approvals. Most *only just* pass the approvals. What would be the point in getting great results at the channel bank, when the phone's own hybrid still gives you a -12dB echo? The phone tends to be worse than the other end. This is why VoIP systems have to use echo cancellation. Hybrids are for little more than keeping the feedback below the oscillation point. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing an outside number -- QUESTION --
On Thu, 2003-11-06 at 18:10, Chris Albertson wrote:
> --- Steve Murphy <[EMAIL PROTECTED]> wrote:
> >
> >
> > I wrote up the following:
> >
> > [outwork]
> > ignorepat => 9
> > exten => _91XX,1,StripMSD,1
> > exten => _1XX,2,Dial,Zap/2/BYEXTENSION
> > exten => _9754,1,StripMSD,1
> > exten => _754,2,Dial,Zap/2/BYEXTENSION
> > exten =>_9NXX,1,StripMSD,1
> > exten =>_NXX,2,Prefix,1307
> > exten => _1307NXX,3,Dial,Zap/2/BYEXTENSION
> >
>
> Try this newer syntax:
>
> [outwork]
> exten => _91XX,1,Dial,Zap/2/${EXTEN:1}
> exten => _9754,1,Dial,Zap/2/${EXTEN:1}
> exten => _9NXX,1,Dial,Zap/2/1307${EXTEN:1}
>
Chris---
Interesting. Can you point to where this is documented? I rooted around
thru the Digium online manual, whitepaper, etc, couldn't find any doc.
While these entries seem to work, I found a small weakness in Asterisk:
It does not recognize my telephone's tones for the number "6". As long
as the outgoing phone numbers do not contain the number 6, I'm OK. In
voicemail, if I dial "6", it doesn't hear it
Can anything be done to improve the recognition of the number 6 in
asterisk? The phone company can recognize it!
murf
> =
> Chris Albertson
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[Asterisk-Users] Sipura SPA-2000 and Asterisk
Hi, I'm using the SPA-2000 with firmware 1.06 on the Asterisk PBX, which works great for taking and placing calls, but for for some reason I can't seem to clear the stutter dialtone by either calling the extension I'm on, or the voicemail system on the Asterisk PBX. If I call my voicemail access extension directly, It tells me I have no messages waiting, yet when I hang up, then pick up the receiver, I still get a stuttered dial tone. Here is the contents of the sip.conf file: [general] port=5060 bindaddr=XXX.XXX.XXX.XXX tos=lowdelay disallow=all allow=ulaw ; ; SIP Entry for sipura line 1 ; This phone is allowed to dial extensions and local phone numbers ; [101] type=friend host=dynamic context=house-local reinvite=no canreinvite=no qualify=300 callerid="Sipura Line 1" <101> mailbox=101 nat=0 ; Sample for sipura line 2 ; This phone is allowed to dial extensions and local phone numbers ; [102] type=friend host=dynamic context=house-local reinvite=no canreinvite=no qualify=300 callerid="Sipura Line 2" <102> mailbox=102 nat=0 Here are my SIP configuration settings for the sipura box from the line1 web page: Proxy: [FQDN of asterisk server] Outbound proxy: [Blank] Register: YES Register Expires: 3600 Use DNS SRV: NO Use Outbound Proxy: NO Use OB Proxy in Dialog: YES Make Call Without Reg: NO Ans Call without Reg: NO Display Name: SIP LINE 1 User ID: 101 Use Auth ID: NO Supplementary Service Subscriptions all set to YES. Audio Config: N/A? Dial Plan: (6x.|1xx|9,911|9,[2-9]xx|9,1[2-9]xx[2-9]xx) Thanks, Steve Rodgers San Diego CA. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Engineers of all kinds can be a bit lax about documentation. However, the documentation police are rightly held in a regard usually reserved for lawyers, realtors, used car salesmen and serial killers. There isn't a single thing to stop anyone that really loves documentation actually producing some. This includes documentation for configuration management. However, I've yet to find a documentation whiner prepared to do anything useful. The ones who are genuine don't whine - they produce something. Regards, Steve Philipp von Klitzing wrote: Hi! 1) No >1.0 release. In fact, no release structure at all really. (Hold your flames: I know this is to be remedied soon, along with backtrack patches for security/stability.) With that comes a "changelog" and some basic documentation. I still find it amazing that "coders" are permitted to add features and introduce patches without ANY kind of documentation. ;-> Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP, Sipura SPA-2000, and Voicemail2
I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why two different directories are being used? The context of the sip phones is not default, it is house-local, and I'm curious as to why the 'local' directory is used, and how it is derived from the SIP phone context. I can correct the problem by making a symlink from local to default, but this does not appear to be the best way to solve this problem. Thanks, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP, Sipura SPA-2000, and Voicemail2
Solution: The context used in voicemail.conf has to match the default context
in sip.conf.
Sip.conf:
[general]
port=5060
bindaddr=192.168.17.2
tos=lowdelay
disallow=all
allow=ulaw
context=default ; Note: this must match voicemail.conf
;
; SIP Entry for sipura line 1
; This phone is allowed to dial extensions and local phone numbers
;
[101]
type=friend
host=dynamic
context=house-toll
reinvite=no
canreinvite=no
qualify=300
secret=x
callerid="Sipura Line 1" <101>
username=101
mailbox=101
nat=0
; Sample for sipura line 2
; This phone is allowed to dial extensions and local phone numbers
;
[102]
type=friend
host=dynamic
context=house-toll
reinvite=no
canreinvite=no
qualify=300
secret=y
callerid="Sipura Line 2" <102>
username=102
Voicemail.conf:
[general]
format=wav
maxmessage=180
[default] ; Note: this was [local]
;
; format: password, name, email address for attached voicemail msgs
;
101 => ,Steve Rodgers,[EMAIL PROTECTED]
102 => ,Karen Rodgers,[EMAIL PROTECTED]
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[Asterisk-Users] Text entry by DTMF
Hi, I've kind of ported a DTMF text extry method I wrote some time ago for Dialogic. It is now a semi-working Asterisk app. I've still got to clean up some stuff in how Festival is used to read back what is entered, and then I think it should be OK. Would anyone here find this useful? It takes an entered string, and returns it as a variable. It might sound clunky, but for short entry of things like names, it actually works quite well. What I am looking for right now is ideas on what people might need in such a facility, to I can tie down the loose ends. Right now it takes: - a variable name to be used when returning the string - some pre-existing entered digits, that might have come from a previous step in a sequence of activities (e.g. if you press a digit at a menu prompt, you might what that digit to be the first digit of the text entry process). - the maximum length of the entered text - an overall timeout, so the entry cannot go on forever. The software uses Festival to read back what is entered. It has numeric, lower case, upper case and symbol modes, and can switch freely between them. The entry technique has been field proven in the Dialogic version. Any useful input much appreciated. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFax question
On Wed, 12 Nov 2003, Freddi Hansen wrote: >Hi, >I am looking at using the softfax that Steve Underwood has developed. >It's very straight forward when you assign an extension for the fax. >A function that several pbx's has is that they listen for the 'faxtone' >for 5 seconds >after 'answer' in the menu where you can enter your local extension number, >it's normally done in parallel with the DTMF detection. I think that You want a fax extension: exten=>fax,1,Blah() A google for 'fax extension' turns up the announcement of this feature here: http://lists.digium.com/pipermail/asterisk-users/2002-October/005414.html ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA 2000 and 404 not found
I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK60fe7596 From: "asterisk" ;tag=as1cf7898d To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.17.6:5060 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK0178ca1c From: "asterisk" ;tag=as6a42fcc6 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.17.6:5061 Sip read: SIP/2.0 404 Not Found To: From: "asterisk" ;tag=as1cf7898d Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK60fe7596 Server: Sipura/SPA2000-1.0.9 Content-Length: 0 8 headers, 0 lines Sip read: SIP/2.0 404 Not Found To: From: "asterisk" ;tag=as6a42fcc6 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK0178ca1c Server: Sipura/SPA2000-1.0.9 Content-Length: 0 8 headers, 0 lines *CLI> sip no debug SIP Debugging Disabled Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=default ; ; SIP Entry for sipura line 1 ; This phone is allowed to dial extensions and local phone numbers ; [101] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=xx callerid="Sipura Line 1" <101> username=101 mailbox=101 ; Sample for sipura line 2 ; This phone is allowed to dial extensions and local phone numbers ; [102] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=yy callerid="Sipura Line 2" <102> username=102 mailbox=102 nat=0 Note that 192.168.17.6:5061 seems to have a problem with "404 not found", wheras 192.168.17.6:5060 does not. Could Asterisk be getting confused about a device with two ports sharing the same IP address? I don't seem to be seeing any traffic being logged from the SPA2000 to Asterisk; it all seems to be going from 192.168.17.2 to 192.168.17.6. If anyone could shed some light on what is going on here it would be sincerely appreciated. Steve Rodgers San Diego, CA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative > please contact me personally by email. Hmmm, this information should be on a website somewhere... -- JustThe.net Internet & New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI> iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI> iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI> iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI> iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI> iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI>
voip2*CLI> iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip2*CLI>
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Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Roger Schreiter wrote: Hmmm, this information should be on a website somewhere... http://www.voipproviderslist.com ? Sure. :) Was just observing that there seems to be a big influx of "We are doing VoIP termination" posts in the past 24 hours and was pointing out - in a roundabout way - that while this is not an inappropriate place to put that information, there are other places where it should go too. -- JustThe.net Internet & New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] box for asterisk
Hi All, We are looking at creating our first asterisk box, what type of server requirements do we need to keep in mind? Is there any preferred server that is used for asterisk? How processor intensive is asterisk? And what is the requirements for the sound card? Thanks, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk solution.
Hi All, We have a Avia Difinity G3R to a Avia EPN connected through 2 dedicated T1 dsics, can I use an Asterisk solution to replace the T1's? If so can anyone give me feedback on what components I should be looking into to help me get started? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX trunk monitoring
I have modified the configuration for dynamic host and registered each
server with the other. The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.
I changed the extensions file to:
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
also tried:
exten => 8475551212,3,Dial(IAX/voip2/${EXTEN},,r)
since voip2 is now a registered user, but it is not trying to call the
other server.
If I leave it as [EMAIL PROTECTED] it works as long as the trunk is up
but it doesn't check to make sure before sending the call there.
Any suggestions?
---
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI> iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI> iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI> iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI> iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI> iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI>
voip2*CLI> iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip2*CLI>
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Re: [Asterisk-Users] Looking for recommendations for home office setups
On Fri, 14 Nov 2003, James Harrell wrote: >We're a small software company, with employees working from home >in three different locations: > - Atlanta: cable modem connection > - Denver: ADSL/PPPOE connection > - Oklahoma City: ADSL/PPPOE connection > >Is this a pipe dream? Here's my goal: > - One phone & one fax at each location > - One telco phone line at each location > - Utilize existing phones, though willing to buy new phones > - Central asterisk server in Atlanta > - Phone line "best rate" routing, outgoing calls routed through > the hard-line at a different location if local, etc. ie: > One can originate a call from the Atlanta phone, have it > routed through the Denver outgoing line to another location > in Denver to achieve a local phone call. > >As far as I understand, this may involve three hardware interfaces, >one at each location (plus a central asterisk box). Each would have: > - TCP/IP connection back to the central asterisk server (perhaps > via a VPN? Or can we just use straight TCP/IP with some form > of authentication. Caveat: we have NAT firewalls at each location. > - Local telco phone line input > - Analog line output for using existing phone, or potentially > go via ethernet to a true IP phone? > >I believe I'm looking for some form of "gateway" box at each location, >controlled by the Asterisk server. Possible? If so, what hardware is >recommended. First, thanks for the very nicely prepared (and well thought-out) message. You are looking for 1 FXO port to bring in the local telco line at each site. You want two FXS ports to provide asterisk dialtone to the existing phone (assuming it's an analog phone) and dialtone to the fax machine. If your solution ends up being an asterisk box at each location, you can do this with 1 X100P and a 2-port TDM400P. Your "best-rate" routing is absolutely no problem. The 'swich' statement will be your friend. You shouldn't need VPN unless you're concerned about encryption. IAX2 (which, as you mention NAT, you'd be -strongly- encouraged to use) will do authentication for you. Good luck, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX trunk monitoring
Creating a separate user and peer does allow me to call over the trunk
using this format:
exten => 8475551212,3,Dial(IAX/voip2/${EXTEN},31,r)
(this must be a bug. Why would the same format not work for a friend?)
It does not solve my original problem of failing the call if the trunk
is down. Calls now are always sent to the voip2 iax user regardless of
whether that user is connected.
Also, voip2 was created as a user and voip2peer was created as a peer.
If I use:
exten => 8475551212,3,Dial(IAX/voip2peer/${EXTEN},31,r)
or
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},31,r)
the call fails as unavailable regardless of whether or not the other
server is running.
The registry information looks wrong too:
voip1*CLI> iax show registry
Host UsernamePerceived Refresh State
209.242.15.34:5036voip1peer60 Request
Sent
209.242.15.34:5036voip160
Rejected
> -Original Message-
> From: Philipp von Klitzing [mailto:[EMAIL PROTECTED]
> aachen.de]
> Sent: Friday, November 14, 2003 4:08 AM
> To: Steve Dolloff
> Subject: RE: [Asterisk-Users] IAX trunk monitoring
>
> You might want to try this:
>
> split the entries for voip1 and voip2, i.e. instead of type=friend
have
> an entry for type=peer and type=user for each of the two machines.
>
> Greetings, Philipp
>
>
I have modified the configuration for dynamic host and registered each
server with the other. The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.
I changed the extensions file to:
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
also tried:
exten => 8475551212,3,Dial(IAX/voip2/${EXTEN},,r)
since voip2 is now a registered user, but it is not trying to call the
other server.
If I leave it as [EMAIL PROTECTED] it works as long as the trunk is up
but it doesn't check to make sure before sending the call there.
Any suggestions?
---
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI> iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI> iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI> iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI> iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI> iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI>
voip2*CLI> iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Fr
Re: [Asterisk-Users] Your thoughts..
Senad Jordanovic wrote: Funny, I am doing the same at the moment... :) We are allowing * to dump call records onto a remote database server. Once there we can do all sort of things with it. My only concern is if this remote server goes down! Use more than one database server... -- JustThe.net Internet & New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA 2000 and 404 not found
John, I talked with Siprura today, and they mentioned that the OPTIONS request must include the USER ID or it'll be rejected with a 404 Not found by the SPA2000. I guess I'll have to spelunk the Asterisk code to see if this is being done. If not, I will request it as a new feature. The only reason I'm concerened about this is that the traffic is making my log files very large and difficult to read. Steve. On Thursday 13 November 2003 22:49, John Todd wrote: > >I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and > > line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the > > asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share > > the same IP address. Every minute I repeatedly get the following output: > > > >SIP Debugging Enabled > >10 headers, 0 lines > >Reliably Transmitting: > >OPTIONS sip:192.168.17.6 SIP/2.0 > >Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK60fe7596 > > From: "asterisk" ;tag=as1cf7898d > > >To: > >Contact: > >Call-ID: [EMAIL PROTECTED] > >CSeq: 102 OPTIONS > >User-Agent: Asterisk PBX > >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > >Content-Length: 0 > > > > > > (no NAT) to 192.168.17.6:5060 > >10 headers, 0 lines > >Reliably Transmitting: > >OPTIONS sip:192.168.17.6:5061 SIP/2.0 > >Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK0178ca1c > > From: "asterisk" ;tag=as6a42fcc6 > > >To: > >Contact: > >Call-ID: [EMAIL PROTECTED] > >CSeq: 102 OPTIONS > >User-Agent: Asterisk PBX > >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > >Content-Length: 0 > > > > > > (no NAT) to 192.168.17.6:5061 > >Sip read: > >SIP/2.0 404 Not Found > >To: > > From: "asterisk" ;tag=as1cf7898d > > >Call-ID: [EMAIL PROTECTED] > >CSeq: 102 OPTIONS > >Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK60fe7596 > >Server: Sipura/SPA2000-1.0.9 > >Content-Length: 0 > > > > > > > >8 headers, 0 lines > >Sip read: > >SIP/2.0 404 Not Found > >To: > > From: "asterisk" ;tag=as6a42fcc6 > > >Call-ID: [EMAIL PROTECTED] > >CSeq: 102 OPTIONS > >Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK0178ca1c > >Server: Sipura/SPA2000-1.0.9 > >Content-Length: 0 > > > > > > > > > >8 headers, 0 lines > > > > > >*CLI> sip no debug > >SIP Debugging Disabled > > > > > >Here's what's in sip.conf: > > > >[general] > >port=5060 > >bindaddr=192.168.17.2 > >tos=lowdelay > >disallow=all > >allow=ulaw > >context=default > >; > > > >; SIP Entry for sipura line 1 > >; This phone is allowed to dial extensions and local phone numbers > >; > >[101] > >type=friend > >host=dynamic > >context=house-toll > >reinvite=no > >canreinvite=no > >qualify=300 > >secret=xx > >callerid="Sipura Line 1" <101> > >username=101 > >mailbox=101 > > > >; Sample for sipura line 2 > >; This phone is allowed to dial extensions and local phone numbers > >; > >[102] > >type=friend > >host=dynamic > >context=house-toll > >reinvite=no > >canreinvite=no > >qualify=300 > >secret=yy > >callerid="Sipura Line 2" <102> > >username=102 > >mailbox=102 > >nat=0 > > > > > >Note that 192.168.17.6:5061 seems to have a problem with "404 not found", > >wheras 192.168.17.6:5060 does not. > > > >Could Asterisk be getting confused about a device with two ports sharing > > the same IP address? I don't seem to be seeing any traffic being logged > > from the SPA2000 to Asterisk; it all seems to be going from 192.168.17.2 > > to 192.168.17.6. If anyone could shed some light on what is going on here > > it would be sincerely appreciated. > > > >Steve Rodgers > >San Diego, CA > > The symptoms are caused by your "qualify=" lines. Every 60 seconds, > an "OPTIONS" request is sent from Asterisk to the destination. I > don't seem to see evidence that supports your statement that packets > to the 5060 port works and 5061 doesn't; it appears that both > requests are sending back the same 404 error. > > This is a bug with the Sipura; actually, more like a missing feature. > I would say "Don't worry about it" since the added traffic keeps your > NAT mappings in place. I'm sure Sipura will come up with a patch if > you and all your friends send in bugnotes to them. > > JT > ___ > 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
[Asterisk-Users] Problem with call pickup -or- what stupid mistake have I made?
For some reason, I can't get call pickup to work between Sip phones or between
Sip and Zap phones. All phones are in the same call group and pickup group
(1). The source code was downloaded and built as of today 11/15/03.
Here's what's in sip.conf:
[general]
port=5060
bindaddr=192.168.17.2
tos=lowdelay
disallow=all
allow=ulaw
context=aliens
;
; SIP Entry for sipura line 1
; This phone is allowed to dial extensions and local phone numbers
;
[101]
type=friend
host=dynamic
context=house-toll
reinvite=no
canreinvite=no
qualify=300
secret=x
callerid="Sipura Line 1" <101>
username=101
callgroup=1
pickupgroup=1
[EMAIL PROTECTED]
nat=0
; Sample for sipura line 2
; This phone is allowed to dial extensions and local phone numbers
;
[102]
type=friend
host=dynamic
context=house-toll
reinvite=no
canreinvite=no
qualify=300
secret=y
callerid="Sipura Line 2" <102>
username=102
callgroup=1
pickupgroup=1
[EMAIL PROTECTED]
nat=0
Here's what's in zapata.conf:
;
; zapata.conf
;
[channels]
;
; 2 ea. X100P plugged into PSTN
;
context=from-pstn-xx
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=no
;callerid=asreceived
group=1
channel=1
;
context=from-pstn-xx
channel=2
;
; TDM100B Port #1 plugged into analog Phone
; This phone is allowed to dial extensions and local and long distance numbers
;
amaflags=documentation
context=house-admin
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=no
threewaycalling=yes
cancallforward=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
relaxdtmf=yes
rxgain=1
txgain=1
immediate=no
musiconhold=default
usecallerid=yes
callerid="TDM100 Zap Phone" <103>
callgroup=1
pickupgroup=1
group=2
channel=3
Here's the verbose output from the console when *8# is dialed on a sip phone:
-- Starting simple switch on 'Zap/3-1'
-- Executing SetVar("Zap/3-1", "ALERT_INFO=Bellcore-r3") in new stack
-- Executing Dial("Zap/3-1", "SIP/[EMAIL PROTECTED]:5060|20") in new stack
-- Called [EMAIL PROTECTED]:5060
-- SIP/192.168.17.6-67bd is ringing
NOTICE[1142127920]: File chan_sip.c, Line 5090 (handle_request): Nothing to
pick up
== Spawn extension (house-admin, 101, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
___
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[Asterisk-Users] SIP soft phone registration
Hello-- I've installed a few X-Lite softphones on windows machines, and am playing with the settings. I've written before about this, and since have discovered that the X-Lite has an option to turn off registration, which I have set, because I bumped into a letter from Mark, saying that Asterisk doesn't do registrations yet. And, without such on the X-Lite, I can still dial out, and get calls. And, best of all, no 60-second repetition of the same error message over and over and over to fill up the logs and the screens. My question: What is this feature? What is SIP registration? murf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk
Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap analogs? In other words, what features will a (more expensive) VOIP phoneset provide, that the analog won't? I know already that asterisk will give me these features with just plain analog phones (&zaptel cards, of course): Voice mail, park & retrieve & MOH, transfer, agents, and a few others. And, if you get an analog with a CID built in, you could have that, too? (Haven't tried that yet). What is the justification for VOIP? just total cost reductions (if the phone is cheap enough?)? Or are there some nicenesses that only VOIPs can supply? murf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com NBX phones
3com made a phone that supported SIP but was discontinued. From what I have heard, the hardware was exactly the same, just a different BIOS. I would like to try or hear of someone trying to flash an NBX phone into a SIP phone. Here is a link to one that sold on Ebay. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3057972164&category=11909 We use an NBX here in my office and the business phones are great. No worry though, 3com's new product VCX7000 ( http://drs.yahoo.com/S=2766679/K=3com+vcx/v=2/SID=e/l=WS1/R=1/H=0/*-http://www.3comvcx.net/ ) line is basically going to be an Asterisk system (plus a large fortune) and will even run Linux (NBX runs VxWorks) Work off of SIP and the phones should be one step up from the current ones. I am sure they will attempt to make everything proprietary but going the direction of SIP is a very good sign. Thanks, Steve Totaro - Original Message - From: "Andrew Nelson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 17, 2003 5:02 PM Subject: [Asterisk-Users] 3Com NBX phones > Has there been any luck getting the 3Com NBX series phones to work with > Asterisk? > > Thanks! > > -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
Re: [Asterisk-Users] Radius on *
looks like critchy is especially bitchy - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 17, 2003 5:40 PM Subject: Re: [Asterisk-Users] Radius on * > Dude, you deserve a double dipping in flaming hot sauce for this stupid > post. > > I'll give you that the question isn't stupid, just the way you went > about asking it. > > First you point out why I hate people who use digest format of a mailing > list that is many to many communications. You just sent the entire > digest to us all when we already had a copy of every message. Worse yet, > you didn't even reference any of the messages in the digest so as making > it all worthless wasted bits. > > Second, you responded to a message and then started a new thread. At > least the digest fiasco you started caused the thread to not connect > anyways, but come on, you see us from time to time make examples of > stupid actions. Why did you have to be that guy? > > AFAIK, no there is no radius accounting built in. You could link to one > with any number of tools through the AGI interface if you so wish. Go > knock yourself out. > > Oh and BTW, UTFG before asking a question. You would have probably seen > that question discussed hear many times before. > > On Mon, 2003-11-17 at 11:04, Sebastian Nocetti wrote: > > Does Asterisk support Radius accounting? > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > 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] problems with alsa (card ac97) in asterisk
critchy is very very bitchy indeed... - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 17, 2003 5:42 PM Subject: Re: [Asterisk-Users] problems with alsa (card ac97) in asterisk > On Mon, 2003-11-17 at 14:41, Dorian Gray wrote: > > Antonio Angel wrote: > > > [chan_alsa.so] => (ALSA Console Channel Driver) > > > asterisk: pcm.c:5486: snd_pcm_sw_params_set_silence_threshold: Assertion `val > > > < pcm->buffer_size' failed.Aborted > > > > > > There is problems between asterisk and alsa or ac97 ? > > > I need any option for to compiler asterisk ? > > > > I guess most people prefer to use oss rather than alsa...? this issue > > was noticed earlier this year, and the fix is now included in cvs: > > http://bugs.digium.com/bug_view_page.php?bug_id=526 > > > > I am also using i8x0 and alsa, and this patch (or latest cvs) works > > great for me. > > Or more likely by the tone of most post here, those of us heavily using > asterisk do not use the soundcard at all. We have better full function > ways of interfacing with asterisk. Phones are not that expensive, and > not hard to come by. Please invest in your education. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > 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] Radius on *
Andrew Kohlsmith wrote: Oh and BTW, UTFG before asking a question. You would have probably seen ... UTFG? May the Google be with you, always :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hold music =]
Hi All, Just installed our very first asterisk system, and we love it! I cant believe the different things you can do with it, just great =] My question is: How do I configure my system to play an mp3 file when a caller gets put on hold? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody using Sphinx
Arnold Ligtvoet wrote: Since I would like the user names to be auto-generated by the system, I would guess that this could best be done using festival with a localized voice. I think there is a Dutch voice for Mbrola with should integrate into festival ( note to self : need bigger harddisk :-) ) Speech recognition accuracy is not great under ideal conditions. Doing what you suggest seems unlikely to achieve any meaningful accuracy. Speech recognition training systems require many occurances of a word or phrase, clearly spoken, before their accuracy becomes useful. A one shot utterance from Festival seems to fail on both counts :-) Bottom line: the very best speech recognition still sucks. As a British speaker I never get more than about 40% accuracy speaking into a US trained recogniser. I have never had better than about 70-80% accuracy on a British trained recogniser. Strangely, my terrible Cantonese gets nearly 100% on SpeechWorks recogniser. :-\ Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody using Sphinx
Anthony Wood wrote: On Wed, Nov 19, 2003 at 10:22:55AM +0800, Steve Underwood wrote: Arnold Ligtvoet wrote: Since I would like the user names to be auto-generated by the system, I would guess that this could best be done using festival with a localized voice. I think there is a Dutch voice for Mbrola with should integrate into festival ( note to self : need bigger harddisk :-) ) Speech recognition accuracy is not great under ideal conditions. Doing what you suggest seems unlikely to achieve any meaningful accuracy. Speech recognition training systems require many occurances of a word or phrase, clearly spoken, before their accuracy becomes useful. A one shot utterance from Festival seems to fail on both counts :-) Sphinx isn't doing general speech recognition, it is determining which of a list of patterns it has you said, like mobile phones do. That is essentially all that any voice recognition currently does. There is little meaningful context directed recognition (a "phrase locked loop" to use an old in joke) in anything available today. So it's fairly easy to tell between "Jennifer" and "Frank" if there are no other options. Many commercial on-line recognisers have serious trouble telling between "yes" and "no" when those are the only two acceptable answers. When you call directory assistance in Australia, the IVR asks you what name you want, and gives you a suggestion out of the top 100 or 200 names, which you can accept or reject. Makes for riducule, but beats waiting on hold. Beware that many of these systems are actually a human operator hiding behind and IVR. I've had people tell me about amazing automated directory enquiry systems in the US, which turn out to be a human masquerading as an IVR. If the list is known to be short, that many not be the case here. Bottom line: the very best speech recognition still sucks. As a British speaker I never get more than about 40% accuracy speaking into a US trained recogniser. I have never had better than about 70-80% accuracy on a British trained recogniser. Strangely, my terrible Cantonese gets nearly 100% on SpeechWorks recogniser. :-\ This is true for general speech recognition, where the computer has a much larger dictionary to match the sound waves against. Only a speaker trained system could even begin to approach these accuracies for general text input. The accuracies I gave are for phone based systems expecting a very limited set of responses from an arbitrary caller. Humans really don't do that much better at raw word recognition, but we heavily apply context to improve things. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hold music =]
Thanks, I tried this and it doesn't play the sample mp3 file when I place a call on hold. I searched around and found an article (http://www.mail-archive.com/[EMAIL PROTECTED]/msg09971.ht ml) that says to download and install mpg123-0.59q-1.i386.rpm which I did and still no luck... any ideas? Thanks, p.s. I have onboard sound which shows as loaded when I do lsmod.... Steve. -Original Message- From: Ing. Angel Gomez [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 5:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] hold music =] Steve Bradwell wrote: >Hi All, > >Just installed our very first asterisk system, and we love it! I cant >believe the different things you can do with it, just great =] > >My question is: How do I configure my system to play an mp3 file when a >caller gets put on hold? > >Thanks in advance, > >Steve. >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > > /usr/src/asterisk/configs/musiconhold.conf.sample and >show application setmusiconhold ___ 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] hold music =]
Thanks, can this happen from a non - sip phone? We have aastra (bell south) 390 analog phones. -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 1:27 PM To: Asterisk-Users Mailing-list Subject: RE: [Asterisk-Users] hold music =] http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat On Wed, 2003-11-19 at 18:53, Steve Bradwell wrote: > Thanks, > > I tried this and it doesn't play the sample mp3 file when I place a call > on hold. I searched around and found an article > (http://www.mail-archive.com/[EMAIL PROTECTED]/msg09971.ht > ml) that says to download and install mpg123-0.59q-1.i386.rpm which I > did and still no luck... any ideas? > > Thanks, p.s. I have onboard sound which shows as loaded when I do > lsmod > > Steve. > > -Original Message- > From: Ing. Angel Gomez [mailto:[EMAIL PROTECTED] > Sent: Tuesday, November 18, 2003 5:02 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] hold music =] > > Steve Bradwell wrote: > > >Hi All, > > > >Just installed our very first asterisk system, and we love it! I cant > >believe the different things you can do with it, just great =] > > > >My question is: How do I configure my system to play an mp3 file when a > >caller gets put on hold? > > > >Thanks in advance, > > > >Steve. > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > /usr/src/asterisk/configs/musiconhold.conf.sample > and > >show application setmusiconhold > > ___ > 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 ___ 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] Cannot do international dial with E1 in Spain
Hi Antonio, This is often a pain with ISDN. What works varies from place to place. Ahm the wonders of standards :-). Setting the dial plan to international is probably right. When you do this you may need to drop the 00 prefix, and start with the country code. Then again, you may not. It varies. Experiment. Its hard to say for sure whether your line has IDD access or not - telco staff are about as trustworthy as used car salesmen and politician. However, most European countries don't seem to offer a non-IDD access option for most lines. Regards, Steve Antonio Castillo Villoslada wrote: Hi, I have a problem with dialling internationals numbers, and I don't now what is the cause. I have one asterisk with a e100p card connected to the Telco (spain/telefonica) and it can dial local and national numbers without problems but when I try to dial a international number it hangs-up. I call the Telco to ask if the E1 can do international calls and it said that it can. I have tried with pridialplan=unknown / local / national / international / private and none of this work. I don't now what to do now, can any one give me a clue of what is happening? The correct prefix is 00 to do international dialling in Spain with E1? -- Attempting call on Zap/g1/0035316694 for [EMAIL PROTECTED]:1 (Retry 1) -- Making new call for cr 37378 Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 4610/0x1202) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] Display (len= 1) [ > Display (len= 1) [ 1> Display (len= 1) [ 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len=17) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316694' ] Sending Complete (len= 0) < Protocol Discriminator: Q.931 (8) len=43 < Call Ref: len= 2 (reference 37378/0x9202) (Terminator) < Message type: STATUS (125) < Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) < Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] < Cause data 0: 01 (1) < Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) < Display (len=28) [ < Display (len=28) [ N< Display (len=28) [ NO< Display (len=28) [ NO < Display (len=28) [ NO E< Display (len=28) [ NO EX< Display (len=28) [ NO EXI< Display (len=28) [ NO EXIS< Display (len=28) [ NO EXIST< Display (len=28) [ NO EXISTE< Display (len=28) [ NO EXISTE < Display (len=28) [ NO EXISTE E< Display (len=28) [ NO EXISTE EL< Display (len=28) [ NO EXISTE ELE< Display (len=28) [ NO EXISTE ELEM< Display (len=28) [ NO EXISTE ELEME< Display (len=28) [ NO EXISTE ELEMEN< Display (len=28) [ NO EXISTE ELEMENT< Display (len=28) [ NO EXISTE ELEMENTO< Display (len=28) [ NO EXISTE ELEMENTO < Display (len=28) [ NO EXISTE ELEMENTO D< Display (len=28) [ NO EXISTE ELEMENTO DE< Display (len=28) [ NO EXISTE ELEMENTO DE < Display (len=28) [ NO EXISTE ELEMENTO DE I< Display (len=28) [ NO EXISTE ELEMENTO DE IN< Display (len=28) [ NO EXISTE ELEMENTO DE INF< Display (len=28) [ NO EXISTE ELEMENTO DE INFO< Display (len=28) [ NO EXISTE ELEMENTO DE INFOR< Display (len=28) [ NO EXISTE ELEMENTO DE INFORM< Display (len=28) [ NO EXISTE ELEMENTO DE INFORM ] -- Processing IE 8 (Cause) -- Processing IE 20 (Call State) -- Processing IE 40 (Display) < Protocol Discriminator: Q.931 (8) len=10 < Call Ref: len= 2 (reference 37378/0x9202) (Terminator) < Message type: CALL PROCEEDING (2) < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 <ChanSel: Reserved < Ext: 1 Coding: 0 Number Specified Channel Type: 3 < Ext: 1 Channel: 3 ] -- Processing IE 24 (Channel Identification) < Protocol Discriminator: Q.931 (8) len=33 < Call Ref: len= 2 (reference 37378/0x9202) (Terminator) < Message type: DISCONNECT (69) < Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) < Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Even
[Asterisk-Users] Stutter Tone all the time?
Hi-- If I have a voicemail box with a number of 1, and say (among other things) in zapata.conf: mailbox = 1 group = 3 context = workext callerid = "Steves Extension"<(999)999-> channel => 6 (assuming I have a TDM 4 port card, and 2 FXO T100P's ) Yes, I really have a mailbox, number 1. From voicemail.conf: [voicemails] 1 => ,Steve,[EMAIL PROTECTED] And, the result is: I get stutter when I pick up the phone. Whether there's voicemail or not. What am I doing wrong? murf signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] can't get caller id?
Steven Critchfield wrote: Callerid in the Us is signaled with FSK between the first and second ring. For more information, you will need to consult the source as it will be definitive. So far, I think it has been established that in the UK, the callerid is sent after a line reversal and before the first ring. The UK uses a similar FSK message format to the US (not identical, but similar). However UK ringing doesn't have a huge silence period in it. Therefore, they send the caller ID before the first ring. Does anyone know why they use a reversal to indicate the CLI is coming? Most digital exchange lines are incapable of reversing the line, so it seems like this choice forces the use of a new line card for any sub. who joins the CLI service - weird! Various alternatives seem simpler. In something like * it would not be necessary to look for the reversal. Conintuously running the FSK receiver isn't that big a compute load. It could be run continuously between calls, and pick up any FSK messages which arrive. I think it has also been established that some locations have callerid as either dtmf or mf. I don't usually pay attention to this as much because I can't help them. DTMF is used in some places. Japan uses FSK, but a rather different message format. There isn't a whole lot of global standardisation in CLI! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] making outside call with sip phone
Hi All, I have a grandstream sip phone which I have figured out and configured to make internal calls. How do I now configure asterisk to allow this phone to make an outside call? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Local numbers to Victorville/Apple Valley, CA
Hey all, I am in the High Desert region of southern California, USA. I was wondering if any of the SIP providers offer numbers serviced out of the following Verizon central offices: Apple Valley (Apple Valley CO/APVYCAXF) Apple Valley (Desert Knolls CO/DSKNCAXF) Victorville (VTVLCAXA) Adelanto (ADLNCAXF) Hesperia (HSPRCAXF) These are the COs which offer prefixes which are local calls from my home. I don't believe there are any facilities-based LECs in this area, which is why I asked about providers colo'ing at the COs listed above, but if there are any facilities-based providers (they'd probably be in Victorville) I'd be willing to hear about them too. Thanks in advance. -- JustThe.net Internet & New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
