Re: [Asterisk-Users] Text to Speech - Someone needs to do this
At 15:41 2003-07-15 -1000, 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. Text-to-Speech (TTS) is usually either formative, created by synthesis of sounds; or concatenative, created by concatenating sounds of actual speech samples. However, concatenative TTS usually works by using small fragments of speech, not entire words. The storage requirements are much smaller, and it gives the system an opportunity to pick units of speech that match the units of speech that precede and follow them. The real trick is to get the correct posidy. Here's three sentences with the same words but each with different prosidy: I said 'yes.' I said yes? _I_ said '_yes_'???!! Both formative and concatenative systems add prosidy. Adding prosidy to whole-word concatentative systems is difficult. If you're in a buying mood, there are some excellent TTS systems available. For example, Rhetorical (http://www.rhetorical.com) has some excellent voices. And they have the funniest TTS current available is the Southern California female voice; I use it for non-serious demos (That's so totally awesome.) Commercial TTS is actually very intelligble and perfectly adequate for many tasks. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ 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
At 10:11 2003-07-16 -0700, Chris Albertson wrote: SNIP 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 emph I /emph said quotequestionword yes /quote/questionword 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. 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. I think we should take any further discussion off-list. Regards, Moshe -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VXML?
At 21:35 2003-07-14 -0400, Kevin Herzig wrote: Anyone know of anybody doing VXML with Asterisk and/or Linux? Open-source VoiceXML (VXML) interpreters are available for Linux. This does you absolutely no good, of course; you must integrate the interpreter into the platform. I.e., input in the interperter is VoiceXML, the output is in some API that must integrate with the platform. Even getting ECMAScript running in VoiceXML is a non-trivial task. Speechworks and Nuance will cheerfully sell you VoiceXML on Linux. I think IBM may also have something... check Alphaworks. I assume that you're attempting to build VoiceXML services on top of SIP? This is something that interests me as well. What architecture do you have in mind, if I may ask? Integrated VoiceXML on Asterisk wouldn't be my initial choice, at least for what I have in mind. Unfortunately, despite a month of futzing around, I cannot get Asterisk to work a softphone -- any softphone -- so I have no practical advice. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2003-06-10 CVS: softphone connection failures
I just downloaded the latest CVS. RTP streams from a X-Lite on a PC, to the Linux box running Asterisk and Linphone, seem to connect: -- Called m4 -- SIP/m4-dd8f is ringing -- SIP/m4-dd8f answered SIP/m12-3649 -- Attempting native bridge of SIP/m12-3649 and SIP/m4-dd8f But then Linphone sends an endless stream of complaints: (linphone:20223): oRTP-WARNING **: Error sending rtp packet: Connection refused. This is new behavior... not to say that X-Lite and Asterisk ever successfully sent streams to each other in my hands, just that this current CVS seems to be a backwards step. SJPhone still has the one-way talk-path problem -- a connection seems to go through, very static-filled indeed, and then after about 5 seconds the talk path goes down. I don't know if it this is a problem with my softphone installation, or if it's an Asterisk problem. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One-way talk paths (without INVITE?) and other issues
I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call from the PC (user m12) to Linphone (usr m4), which rings * I answer on Linphone * Asterisk attempts to set up a talk path. Here's the output from Asterisk, with Linphone (m4) connecting to the PC (m12): -- Called m4 -- SIP/m4-8f2b is ringing -- Registered SIP 'm12' at 172.28.54.34 port 5060 expires 500 -- SIP/m4-8f2b answered SIP/m12-195f -- Attempting native bridge of SIP/m12-195f and SIP/m4-8f2b I don't know how the registered statement appreared in the middle. At this point I can talk into the PC and hear it on Linphone -- but I cannot speak into Linphone and hear myself on the PC. After about 10 seconds, possibly less: * The PC phone gets a hangup message (BYE). * Linphone does *not* get a hangup message and remains offhook. Any attempt to call Linphone from the PC results in Asterisk routing the call to voicemail. I must manually hang up Linphone. Oddly enough, the caller ID displayed by Linphone, and apparently sent by Asterisk, is incorrect. Instead of showing m12 as the caller ID, Linphone receives m1 as the caller ID: INVITE sip:[EMAIL PROTECTED]:5062 SIP/2.0 Via: SIP/2.0/UDP x1.x2.x3.x4:5060;branch=z9hG4bK0c4d7e4c From: m1 sip:[EMAIL PROTECTED];tag=as62b91e33 To: sip:[EMAIL PROTECTED]:5062;tag=4210403538 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 159 (x1.x2.x3.x4 substituted for the actual IP address.) To my fairly untrained eye, this looks like a legitimate proxy message but the caller ID is wrong. My SIP configuration file does contain both m1 and m12 as legitimate callers: [m1] type=friend username=m1 host=dynamic permit=x1.x2.x3.0/16 [m12] type=friend username=m1 host=dynamic permit=x1.x2.x3.0/16 I also note the following Asterisk warnings. I cannot tell if they happen just before or just after I lose the one-way voice path: WARNING[81926]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) WARNING[81926]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) Furthermore, I have yet to see a SIP channel disappear after a call is over. They are always listed as active, even hours later. Here are is the result of show sip channels: Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 172.28.54.160m4 478c64565be 00104/0 0ms ms 0 172.28.54.160m4 4d330ced01e 00104/1 0ms ms 0 172.28.54.160m4 0bb7855f15b 00104/1 0ms ms 0 172.28.54.160m4 3db8538b4b4 00104/1 0ms ms 0 4 active SIP channel(s) but none of these calls are active in any sense of the term that I can think of. I have tried to use the sip show channel command for further testing, but apparently I don't understand the syntax of the command -- sip show channel 478c64565be and sip show channel m4/478c64565be and sip show channel SIP/m4-8f2b all give the same error message, no such SIP call ID 'whatever'. Either I don't understand the sip show channels command, or there's a bug. My questions are: * How is it that I get one-way voice paths? Is this a configuration problem? Are the INVITES not getting through but the voice paths established anyway? * What's the problem with the incorrect caller IDs? I have *no* caller ID settings that I'm aware of in my *.conf files. The PC's program registers as m12 but Asterisk sends m1 as the name. PC-side debugging shows that the PC sends [EMAIL PROTECTED] as its name. Although this feels like a bug, I strongly suspect that I'm missing some simple SIP configuration issue, but I haven't been able to track it down just yet. And I'd like to clarify any other issues before starting on a bug hunt. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
At 22:10 2003-07-01 -0400, you wrote: To find out what version yuor using, dial *999 and a debug/trace window will appear. In the SIP messages it will indicate the type of UA your using and the version. example below: try another call attempt with this window open and capture the call flow and send it to me. See below in bold or (User-Agent: X-PRO build 1035). Mike Mike, Thanks for writing. After you wrote, I had a brainwave and remembered that the UA is in the debug logs on asterisk... Be that as it may, I'm using X-Lite, and the latest version from their web site is build 1016. That's as of today (Wednesday, 07-02), and that's the version that I'm using on my system. I will send you the debug logs off-list, but here's the question: is it likely that X-Pro works but X-Lite still has trouble with its SDP? Regards, Moshe - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 9:32 PM Subject: Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite At 20:08 2003-07-01 -0400, Michael Kane wrote: What version of X-Lite are you using. The latest is build v1035. There where problems in earlier releases with SDP values, that could be the reason you not seeing invites or media. I had issues only with the media not setting as X-lite tried to negotiate media with another endpoint and teh SDP was hosed. Mike Mike, I downloaded the version I'm using late last week or early this week. It ought to be the latest. There's no way to tell from looking at the app (that I can find) what build it is. Let's see... created June 18th. That's pretty recent. I think it may be bug-report time. Any softphones you recommend for PC or for Linux? I'm actually rather disappointed with everything I've tested, with the exception of SJPhone (but I've only fiddled with it very briefly). Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 7:24 PM Subject: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite Today's frustrated programmer award goes to Linphone, which has the following debug output: (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that this is a linphone problem... other clients don't report problems. In other news, according to my trace of Ethernet packets, the PC softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my Linux softphone, nor does it play out the UDP packets that it receives. This is not an asterisk problem because the PC's SJphone does work -- sortof. The PC's SJPhone does send/receive packets directly to asterisk. But there seems to be a problem with someone's negotiation protocol -- Kphone seems to expect GSM and SJPhone is apparently sending G.711. You can imagine how that sounds. More later if I get it straightened out. -- Moshe Yudkowsky * http://www.Disaggregate.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 -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ 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 -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sorry 'bout that
Sorry 'bout that vacation message. Procmail usually is smart enough to avoid sending replies to mailing lists. I put in a rule to prevent this from happening again. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Today's frustrated programmer award goes to Linphone, which has the following debug output: (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that this is a linphone problem... other clients don't report problems. In other news, according to my trace of Ethernet packets, the PC softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my Linux softphone, nor does it play out the UDP packets that it receives. This is not an asterisk problem because the PC's SJphone does work -- sortof. The PC's SJPhone does send/receive packets directly to asterisk. But there seems to be a problem with someone's negotiation protocol -- Kphone seems to expect GSM and SJPhone is apparently sending G.711. You can imagine how that sounds. More later if I get it straightened out. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
At 20:08 2003-07-01 -0400, Michael Kane wrote: What version of X-Lite are you using. The latest is build v1035. There where problems in earlier releases with SDP values, that could be the reason you not seeing invites or media. I had issues only with the media not setting as X-lite tried to negotiate media with another endpoint and teh SDP was hosed. Mike Mike, I downloaded the version I'm using late last week or early this week. It ought to be the latest. There's no way to tell from looking at the app (that I can find) what build it is. Let's see... created June 18th. That's pretty recent. I think it may be bug-report time. Any softphones you recommend for PC or for Linux? I'm actually rather disappointed with everything I've tested, with the exception of SJPhone (but I've only fiddled with it very briefly). Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 7:24 PM Subject: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite Today's frustrated programmer award goes to Linphone, which has the following debug output: (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that this is a linphone problem... other clients don't report problems. In other news, according to my trace of Ethernet packets, the PC softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my Linux softphone, nor does it play out the UDP packets that it receives. This is not an asterisk problem because the PC's SJphone does work -- sortof. The PC's SJPhone does send/receive packets directly to asterisk. But there seems to be a problem with someone's negotiation protocol -- Kphone seems to expect GSM and SJPhone is apparently sending G.711. You can imagine how that sounds. More later if I get it straightened out. -- Moshe Yudkowsky * http://www.Disaggregate.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 -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connections, but no voice paths except by console
I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems to work reliably. Kphone will register and connect, but if I dial a different softphone on the same machine and get routed to voice mail, I hear the voice mail announcement but I am unable to leave a voice mail messsage. The reason is explicit on the debug output (I nearly used the phrase ROP!), which says that the voicemail SIP connection is dropped because there's been no input. If I reach the other softphone, I cannot transmit from one to the other. Is the inability to transmit part of the continuing problems between Asterisk and ALSA? (2) If I dial from the console into voicemail, I can hear and transmit. If I dial to/from the console to a softphone, then I can transmit audio from the console to the softphone (apparently -- there's no way I can see to debug this to determine who is getting audio from whom). (3) Softphones on a different PC: Using X-Lite from my (*shudder*) Windows box, I can connect to the console on my PBX, or be routed to voicemail. In either case, I cannot transmit audio in *either* direction, from the PC to the console, or the console to the PC. E.g., the softphone does not hear the audio output of the voicemail announcement. If I let the PC's softphone go to voicemaill, the debug output shows that * drops the call because there's no audio. I'm deducing that for some reason I an not routing *any* audio via SIP. Is there some configuration issue I'm missing? -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connections, but no voice paths except by console
So, as usual, about 30 seconds after I send off the message I come to the realization that I've done a Dumb User Trick (aka operator error). * noload of chan_oss.so chan_alsa.so does disable the console. * I can call from a softphone on the PBX to the PBX's voicemail and transmit audio! * But -- I cannot transmit or receive audio from a softphone on a different machine. I can connect, but there's no audio sent or recevied to the PC. They're all behind the same firewall, the debug output shows that the RTP port is correct. Any ideas for something else simple I've missed? At 12:16 2003-06-30 -0500, Steven Critchfield wrote: If you have the console working, then you can't use a softphone on the same machine. If you want to test with a softphone, set the console driver to noload and try again. This would probably be the same for your dialing to other SIP phones. I'm guessing that ALSA doesn't allow more than one connection to the mic and asterisk already has it. Alsa I think allows multiple connections to the speakers though. On Mon, 2003-06-30 at 11:28, Moshe Yudkowsky wrote: I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems to work reliably. Kphone will register and connect, but if I dial a different softphone on the same machine and get routed to voice mail, I hear the voice mail announcement but I am unable to leave a voice mail messsage. The reason is explicit on the debug output (I nearly used the phrase ROP!), which says that the voicemail SIP connection is dropped because there's been no input. If I reach the other softphone, I cannot transmit from one to the other. Is the inability to transmit part of the continuing problems between Asterisk and ALSA? (2) If I dial from the console into voicemail, I can hear and transmit. If I dial to/from the console to a softphone, then I can transmit audio from the console to the softphone (apparently -- there's no way I can see to debug this to determine who is getting audio from whom). (3) Softphones on a different PC: Using X-Lite from my (*shudder*) Windows box, I can connect to the console on my PBX, or be routed to voicemail. In either case, I cannot transmit audio in *either* direction, from the PC to the console, or the console to the PC. E.g., the softphone does not hear the audio output of the voicemail announcement. If I let the PC's softphone go to voicemaill, the debug output shows that * drops the call because there's no audio. I'm deducing that for some reason I an not routing *any* audio via SIP. Is there some configuration issue I'm missing? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk ALSA module not working
I've made a dumb user error: a typo in modules.conf had me loading both chan_oss.so and chan_alsa.so. chan_alsa.so works. chan_oss.so works. But you must pick one or the other. I will submit a patch in a few hours for moduules.conf so that Asterisk will work out-of-the-box. Wade, if you're not already doing so, use the command asterisk -d and look at the output; I use asterisk -d | tee /tmp/asterisktrace so that I will have a record to look at. At 15:08 2003-06-24 -0400, you wrote: I have exactly the same problem. RedHat 8 w/latest Alsa. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Moshe Yudkowsky Sent: Tuesday, June 24, 2003 2:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk ALSA module not working Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working Clients for Linux?
All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is trying to connect. LINPHONE -- Linphone does not work with ALSA, nor with ALSA's OSS emulation. I've used pre-packaged version of Linphone and compiled my own, using --enable-alsa to get it up and running. NOTE -- not all versions of ALSA let you do this! There's a silent failure to compile -- check config.log to make certain that the ASOUND_H is defined. GNOPHONE -- Gnophone also does not work with ALSA, although I haven't yet tried to compile it from a tarball/CVS. With ALSA version 0.9.4, GnoPhone won't even give me a chance to configure the sound card. IIRC GnoPhone did work with 0.9.2 and possibly 0.9.3. 1. I invite comments from people who can get these clients to work with ALSA 0.9.4. I am working with the latest CVS of ALSA. 2. Are there other open-source clients that I've missed? Regards, Moshe P.S. I've fiddled with the CVS versions only enough to get things to compile. E.g., in linphone, configure.in must be changed to find alsa/asoundlib.h instead of sys/asoundlib.h. Well, ok, I did add -mfmath=sse to some of the Makefiles but that should be harmless enough... -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ALSA module not working
Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or chan_alsa.so; as of this morning, only chan_oss.so will load. The symptoms are very straightforward: If chan_alsa.so loads, no subsequent module loads. An outward sign is that show dialplan will have just a few items and not the entire dialplan because pbx_config.so doesn't load, and important functions like stop now do not exist at the console. There are no error messages, and just the ordinary debug messages. I haven't had any success deciphering the problem -- I'm still figuring out its extend and ramping up on new sections of Asterisk code. I am using the latest (2003-06-24) CVS of asterisk. Question: Is anyone using Asterisk's chan_alsa.so and ALSA version 0.9.4 from a later CVS? I like to know if it's some problem with my local configuration or a bug I haven't found. Related question: Are OSS and ALSA mutually incompatible? I would think so, but there's nothing in the documentation. If they should *not* be loaded simultaneously, I will submit a few lines to patch modules.conf. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Applications, dialplan not loading
Just a brief progress report on the the applications and dialplan not loading: If I don't load chan_alsa.so, by using noload=chan_alsa.so in modules.conf, I do get the dialplan, apps, and etc. (I received a hint offlist from someone who had problems who'd tried a different version of this solution.) I suspect that the problem is a conflict between the libasound2 libraries in the Debian package and the libasound provided by the latest version of ALSA. I am working the issue. Problems to solve: * Resolving library issues * Determining why asterisk does not issue sufficiently complaints about chan_alsa.so (or whatever it is that's blocking loading the dialplan.) -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .gsm files
At 10:34 2003-06-15 -0400, you wrote: Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The sox program will convert wav into gsm: $ sox foo.wav foo.gsm does the trick. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Applications, dialplan not loading
At 18:31 2003-06-15 +0200, you wrote: But you're using a packaged version of asterisk? Have you tried with downloading, compiling and installing from cvs? The pre-packaged .deb always gave me this problem -- stop now never worked, but at least it loaded all the dialplans when I'd start it. The CVS version used to work, with all functions enabled including stop now. However, The CVS no longer works. I tried the latest CVS as of today (06-15). I'm still experimenting... -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Applications, dialplan not loading
I checked that; the directories point to the correct places. As I mentioned in a follow-up message, strace shows that my system doesn't even try to load the modules. I can load app_playback.so, etc., followed by load pbx_config.so to get a dialplan -- but then the system doesn't find the voice files, and doesn't even try to find the voice files. autoload=yes in /etc/asterisk/modules.conf. I welcome any suggestions. Because of the Sabbath I haven't had an opportnity to figure out how to attach ddd to the correct aseterisk process to debugh file.c. I am using the CVS of 06-13 (Friday). At 17:03 2003-06-14 -0500, you wrote: Is it possible your asterisk.conf reflects different directories for storing your modules? Mark On Fri, 13 Jun 2003, Moshe Yudkowsky wrote: At 09:32 2003-06-13 -0500, you wrote: Since you're using the sound card for testing you need to change in the /etc/asterisk/alsa.conf or oss.conf context=local to context=default Martin, Thanks very much for the suggestion. I tried setting context=default in alsa.conf. That didn't work. I decided to delete oss.conf to see if it made a difference, and that didn't work either. Neither did re-creating oss.conf and using context=default. I should mention that dial [EMAIL PROTECTED] didn't work either; not surprising, since there's no dialplan loaded other than the parkedcalls context. I'd appreciate any further suggestions! -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.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 -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Applications, dialplan not loading
I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I have a complete system installed with all the demo packages. As I have done previously, I attempted to test the system using dial 500. This failed with the messsage No such extension '500' in context 'local'. I have diagnosed several problems: (1) None of the application modules seem to autoload, so Macro and Zapateller are not loaded automatically. (/etc/modules.conf does contain autoload=yes). I can load these from the command line, but they do not autoload as they did in the past. (2) Other built-ins do not load. E.g., stop now is missing when I type help, and 'asterisk -rx 'stop now' fails with No such command 'stop' (type 'help' for help). (3) The dialplan does not load. Using show dialplans, I expect to see the demo and default dialplans as given in the sample extensions.conf; however, the only dialplan I see is the parkedcall dialplan. The log file does not shed any light (to my eyes, at least) as to why the dialplan in extensions.conf does not load. What I find extremely puzzling is that this worked with a previous CVS, the one I first got on about 06-09. Since then I have upgraded to the latest version of alsa, but I can't understand why that would make a difference. (Although the log file complains about chan_oss and the sound resouce being busy, it does load alsa and dialing a parked-call extension does execute correctly). Comments? Suggestions? -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabled echo canceller because of tone (rx)
At 10:30 2003-06-13 -0400, John Congdon wrote: Does anyone know what this means? It is in DMESG, and we have people complaining about echo. Disabled echo canceller because of tone (rx) Guess: to prevent multiple echo cancellers from working on the same call, an echo canceller adds a tone -- was it 100 Hz? -- to the voice path. Other echo cancellers, after detecting the tone, will stop. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Applications, dialplan not loading
At 09:32 2003-06-13 -0500, you wrote: Since you're using the sound card for testing you need to change in the /etc/asterisk/alsa.conf or oss.conf context=local to context=default Martin, Thanks very much for the suggestion. I tried setting context=default in alsa.conf. That didn't work. I decided to delete oss.conf to see if it made a difference, and that didn't work either. Neither did re-creating oss.conf and using context=default. I should mention that dial [EMAIL PROTECTED] didn't work either; not surprising, since there's no dialplan loaded other than the parkedcalls context. I'd appreciate any further suggestions! -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Disabled echo canceller because of tone (rx)
At 10:04 2003-06-13 -0600, you wrote: But we get this message all the time though on normal voice calls and it's not because we're calling fax machines or modems. The tone is put in by long-distance voice network echo cancellers -- calls over 400 miles IIRC. This prevents multiple ECs from trying to echo-cancel the conversations. So it's not just a fax/modem tone. It's a Bell-standard telephone-network EC tone. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Applications, dialplan not loading
Some more info on this problem: (1) None of the application modules seem to autoload, so Macro and Zapateller are not loaded automatically. (/etc/modules.conf does contain autoload=yes). I can load these from the command line, but they do not autoload as they did in the past. (2) Other built-ins do not load. E.g., stop now is missing when I type help, and 'asterisk -rx 'stop now' fails with No such command 'stop' (type 'help' for help). (3) The dialplan does not load. Using show dialplans, I expect to see the demo and default dialplans as given in the sample extensions.conf; however, the only dialplan I see is the parkedcall dialplan. If I load pbx_config.so at the cli, then load apps_playback.so and apps_macro.so, I can eventually get the dialplan loaded. This is problematical to say the least! What I also find is that dial 500 fails because the demo file, in the demo directory, cannot be found -- it's printed out as a WARNING in verbose mode. However, the file is very clearly in the directory and is present. I suspect, but cannot prove, there's hardcoded path lurking in the *.c files somwhere... unless someone has a suggestion? -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
