[Asterisk-Users] Re: IP Softphone Recommendations
Anton Krall [EMAIL PROTECTED] writes: Do you know if any of these clients like xlite or firefly could be preconfigured perior to deployment or maybe customized with a background image or skin? Well, if you think of specific prefs settings or options as preconfiguration, at least with XLite/Linux this should be possible. More precisely, the XLite state is saved completely into $HOME/.Xscrc So defining a configuration and then deploying XLite together with that file should do what you want. Don't know about Windows though. Maybe registry hacking or something. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: astrecipes v2.0
lenz [EMAIL PROTECTED] writes: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. Thanks l. Good idea, but don't we have already the Wiki tips/hints, editable by anybody ? I understand people like to contribute, which is great. But spreading the info all over the web instead of centralizing it might be not so a great. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Jesse Guardiani [EMAIL PROTECTED] writes: I don't know about X-Lite, but sjphone seems only to support OSS. One of my requirements is ALSA support. Thus linphone and gnomemeeting. But, interestingly, gnomemeeting seems to be the only client capable of full duplex audio using ALSA+DMIX+DSNOOP+ASYM. Ah, I remember a thread about that on the GM list a couple of weeks ago, so that was you I presume. Well, XLite is OSS too, afaik, so that probably wouldn't help you either. Anway, pushing for an GM alpha snapshot with SIP support might still be an option compared to going through the H323 pile. Damien promised me twice http://mail.gnome.org/archives/gnomemeeting-list/2005-February/msg00018.html http://mail.gnome.org/archives/gnomemeeting-list/2005-April/msg00069.html to produce something workable, i.e. a release 1.3.1 as per the last mail. So if you reminded him too that at least some people are waiting for GM SIP support, it might accelerate the process a bit :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Brian Capouch [EMAIL PROTECTED] writes: Bruno Hertz wrote: Jesse Guardiani [EMAIL PROTECTED] writes: I don't know about X-Lite, but sjphone seems only to support OSS. One of my requirements is ALSA support. Thus linphone and gnomemeeting. But, interestingly, gnomemeeting seems to be the only client capable of full duplex audio using ALSA+DMIX+DSNOOP+ASYM. Ah, I remember a thread about that on the GM list a couple of weeks ago, so that was you I presume. Well, XLite is OSS too, afaik, so that probably wouldn't help you either. xlite works OK with the OSS emulation for Alsa. Sure. I felt though the main trouble spot was asym properly working with the OSS emu (dmix and dsnoop apparently do). If you could confirm it does, all OSS only softphones would of course be candidates given the above requirements. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Unbelievable...
Rich Adamson [EMAIL PROTECTED] writes: As only one individual, I thought their statements were very straight- forward and clear. Having worked as a senior manager in a technical organization, a large number of tehcnical people simply do not comprehend some words (or read other words into whatever they happen to be reading), or, jump to conclusions based on their technical knowledge that are unreasonable (contractually or otherwise). The wording is very obviously oriented toward those types, and I'd bet a fair amount they _still_ receive calls that are clearly answered on their web site. Regardless of what their web site says, they've provided me with the best service of the half dozen itsp's that I've worked with directly. And, I don't work for them or represent them. Interesting you say that, since I thought their statement wasn't that offensive, but rather looked like a fairly emotional reaction to the severe pressure they might experience right now, and which, as they say, apparently starts comsuming resources better spent on trouble shooting. Especially, those of us who have already worked in some kind of online business will recognize the situation and mood they apparently are in, and how unpleasant it can be. Although, on the other hand, a pissed off customer understandably might have a hard time feeling compassionate. Anyway, I think that just because ppl take money for service doesn't necessarily obligate them to take any shit customers might come up with as well. It's the service which is paid for, so if it isn't delivered for whatever reasons, all one basically is entitled to is getting the money back and maybe compensation, depending on the type of service and contract. Also, it's clear whom they are addressing in that statement, i.e. those people who continue mounting pressure on them through various channels in a counterproductive and 'abusive' fashion, some of them maybe really just to 'vent frustrations'. Well, if so, why not let them do their venting in that particular direction as well and move on to the real issues ... ? Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Jesse Guardiani [EMAIL PROTECTED] writes: Wait a sec... COME TO THINK OF IT! Why not run asterisk on your linux box that you are running GnomeMeeting on, and use it to convert between H.323 and IAX and SIP??? After all, it is a penguin... That's certainly a good alternative. I'm currently in the process of hacking up the latest linphone (1.0.1) to fix a few personal show-stoppers. If I can get it to the point that I like it, then I'll probably just go with linphone. But you're right. If it's took much work, then I'll probably just start running asterisk on my laptop to do H.323 to SIP conversions. Thanks for the suggestion! I hadn't thought of that yet. I'd been looking at things like the commercial sip323 program, but I hadn't thought of doing it with a local copy of asterisk. If your only reason to stick to H323 is Gnomemeeting you could try other softphones as well. Especially, the XLite beta for Linux looks promising, and some people like SJphone for Linux. Also, SIP support for Gnomemeeting is underway, but development is slow. I'm constantly pointing out to them how much interest there is, but things still seem to take their time ... Finally, on a recent discussion about the future design of GM on their list, I was surprised to learn that quite a few people really use it for direct PC to PC video calls over the internet. So somehow, after extensive NAT and router fiddling I guess, direct calls apparently work even with H323 (there is already support built into GM for external IP address discovery, as you know, so those remarks about transmission of bogus IP addresses on H323 level probably don't really apply in this case). Anyway, I myself use the setup recommended above, i.e. local * server as protocol translator, and it works reasonably well. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail
Sig Lange [EMAIL PROTECTED] writes: Starting around Apr 14th Gmail has started marking all messages for Asterisk-Users as spam. Prior to that on google groups someone created a asterisk-test group (seperate from this group). Is this perhaps related? I believe it all has happened within a week time frame. Gmail is a great service but if this is what's going to happen I will quit using gmail. I'm giving a shot out to see any other gmail users out there having this problem. My Asterisk-Dev seems to be unaffected. Who's having similiar issues? Slightly OT, but for those of you who don't want high volume lists clutter their mailboxes anyway, let me remind you that gmane.org provides a (standalone) news gateway to many mailing lists, including the * ones. If you use that gateway, you can read and post to this list via a newsreader. You need to stay subscribed though to those lists which require subscriptions, but you then can disable mail delivery to your inbox on your respective list options page. Also, upon first time posting to a particular list, gmane will take you through a otherwise painless registration procedure, as detailed on their website. Advantages: * pull semantics (news) vs. push (mail) * especially no more cluttering or drowning of your mail inbox * also, lightens the load on the digium list server(s) * no more unconfigurable spam filters bullshitting you (gamil). Disadvantages: * initial setup * those who don't know how to deal with news/usenet might have a difficult time using it. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Running asterisk without special hardware
Manish Sapariya [EMAIL PROTECTED] writes: Hi, I was going through some of the list postings...and I felt like if want to do voip within a LAN, I might have to install Asterisk on every machine. I hope it is not the case. What I understand is (or what I want is) - Install asterisk on one of the machine on LAN - Install softphone on all the machines who want to participate in voip communication - Configure softphones to use the asterisk as the service provider (I dont know how to do this...will figure out from the softphone i use) Please correct me if I misunderstood something. Thanks, Manish Well, add to that the configuration of your (single) asterisk installation itself, which basically means setting up a dial plan, and you got the right picture. Note: extensive documentation on practically all aspects of * configuration can be found on the Wiki http://www.voip-info.org/wiki-Asterisk Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Running asterisk without special hardware
Damian Funnell [EMAIL PROTECTED] writes: Hi Manish, Sure can, although you will need a timing source. Not necessarily. In a pure VoIP environment, I don't know of any asterisk application which needs timing other than meetme. I.e. if you need conferencing, you'll need ztdummy as a timing source. If not, you can just download * 'as is', compile and install it into some place, and finally set up your dial plan. That's it. Please read the Wiki for details on * setup and ztdummy/timing as well. All this info is readily available there, and in detail, too. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: From OH323 to SIP or OH323 without gatekeeper
Guillermo Salas M. [EMAIL PROTECTED] writes: Bruno Hertz wrote: Joe S [EMAIL PROTECTED] writes: Hi, I am new with asterisk. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocol without having a gatekeeper. I can make a call from SIP to OH323 by specifying it in the extensions.conf file, like: exten=1001, 1, Dial(OH323/10.10.10.1) so I was wondering if there was a way to call from OH323 to SIP or OH323. Sure. Just specify in oh323.conf the context where incoming calls should go. That context then can include dial statements for any protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to setup dial plans. Finally, instruct your H323 phone to use asterisk as a gateway resp. proxy, not a gatekeeper. Any calls will then go through asterisk, and to the context you specified. I'm doing that with Gnomemeeting all the time, and it works without problems. Mayabe can you show us a little sample? I can call from Gnomemeeting to Xlite, but no from xlite to gnomemeeting. Well, the direction GM - XLite basically was what we were talking about. For the other direction, i.e. calling an H323 client without gatekeeper, you simply dial the IP address or domain of the client, like Dial(OH323/yourclient.yourdomain.com:1720) or Dial(OH323/192.168.0.123:1720) somewhere in your Dialplan. E.g. if you want to do XLite - GM, such a dial statement should be part of the context into which your incoming SIP calls are routed, as specified in sip.conf. Example: * sip.conf context=default * extensions.conf [default] exten = 123,1,Dial(OH323/192.168.0.123:1720) I.e. dialing '123' with XLite registered on your server would in this case result in calling a hopefully running H323 client on IP address 192.168.0.123. Of course, if your H323 clients use dialup connections, setting up a dial plan for them without using a gatekeeper may prove to be troublesome. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper
Joe S [EMAIL PROTECTED] writes: Hi, I am new with asterisk. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocol without having a gatekeeper. I can make a call from SIP to OH323 by specifying it in the extensions.conf file, like: exten=1001, 1, Dial(OH323/10.10.10.1) so I was wondering if there was a way to call from OH323 to SIP or OH323. Sure. Just specify in oh323.conf the context where incoming calls should go. That context then can include dial statements for any protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to setup dial plans. Finally, instruct your H323 phone to use asterisk as a gateway resp. proxy, not a gatekeeper. Any calls will then go through asterisk, and to the context you specified. I'm doing that with Gnomemeeting all the time, and it works without problems. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper
Joe S [EMAIL PROTECTED] writes: Hi Bruno, Thanks for the input, one question. Let's say I define context=default in my oh323.conf. Then, in my extensiions.conf I have: [default] exten=1002, 1, Dial(SIP/1002); 1001 is an Xlite SIP UA so how do I call a sip user like from NetMeeting, is it like 1002@ip_address_of_gateway?? Argh, this is really a netmeeting issue. Remember I said 'point your phone to use asterisk as proxy/gateway'? Now, the question is whether your client is smart enough to allow that, and if so, how it's done. I.e. in GM I can set the proxy in the preferences dialog, and then just dial 1002 with your above example. Now, I don't use netmeeting myself (and have no Windows installed, for that matter), but a colleague of mine tells me it should be configurable via Tools-Options-General-Advanced Calling. So try setting the gateway there, and if it's configurable simply dialing 1002 should suffice. If not, I'm afraid Google resp. MS support might be the only friends left in this matter. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper
Joe S [EMAIL PROTECTED] writes: Hi Bruno, Thanks I appreciate your help its really working, I just dial 1002 for NM, and Xlite is ringing. Joe. Welcome. Thanks for your feedback, too. Good to hear it works, especially if similar questions come up in the future. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reply-To?
Jean-Michel Hiver [EMAIL PROTECTED] writes: Jean-Michel Hiver wrote: Oops, sorry for the list reply :/ Actually, why does the Reply-To point to the Asterisk Users mailing list? This breaks the reply to sender only / reply to all / list reply functionality of my mailer. It's really broken :( Some would say your mail client is broken. What you're complaining about is generally called 'reply-to munging', and there's been a long discussion about this. Google reveals more, like these two oppositional opinions http://www.unicom.com/pw/reply-to-harmful.html http://www.metasystema.net/essays/reply-to.mhtml Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep [EMAIL PROTECTED] writes: http://groups-beta.google.com/group/Asterisk-test Stuff shows up fast! Anyone have insight on this, did I miss something? Apparently, somebody created that group on google groups and subscribed it to the * mailing list. As long as registered, anybody can do that. This does afaik not imply that those groups will show up on news servers, like e.g. the Debian moderated groups which just mirror their mailing lists, and to which posting isn't possible either, btw., because they're mirrored as moderated groups. So the whole thing lives on google only, and it's real (and probably only) benefit is the search capability. Which is still useful enough, though :) I didn't find out yet how long google will keep the postings. Maybe 'indefinitely', as they generally seem to do with newsgroups, maybe just a limited time ... Looks like the goup was created around end of Feb, beginning of March. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] writes: a couple other lists that I am on got notices last night that they were added to google groups. I wonder if this is a google marketing ploy, seek out all lists and subscribe them then spam the various lists informing the individuals that instead of seeing it free in your email box you can make google money by using a web browser and watching ads. May be. Subscription options for those groups however include getting new articles by mail. Didn't check that out though, so the mails themselves might contain ads either. What I'm still wondering about is, while you can post to that group, whether your postings are actually propagated to this list. Did anybody try that? Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep [EMAIL PROTECTED] writes: What I'm still wondering about is, while you can post to that group, whether your postings are actually propagated to this list. Did anybody try that? Regards, Bruno. Postings to google are not mirrored here, tried it. I think we are going to start seeing many people new to * using the google group and not getting the benefit of the infinite wisdom here. I can not imagine how you would sync them, that would only result in a circular posting nightmare. Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. Don't know how mailman does the auth here, but assumed it's done on the envelope sender google could even post to this list under their registered user and still maintain the From of the original poster, who after all registers on google with an email address. As I see it, there's no technical issue which could prevent google making their (mailing list) groups working 'proxies' to the real lists. Actually, I would be surprised if that wasn't their goal, as they seem to have a tendency lately to suck up anything related to internet communications. Usenet was first, mail is in the works, and as others have speculated voip might be next. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reply-To?
John Novack [EMAIL PROTECTED] writes: Bruno Hertz wrote: Jean-Michel Hiver [EMAIL PROTECTED] writes: Jean-Michel Hiver wrote: Oops, sorry for the list reply :/ Actually, why does the Reply-To point to the Asterisk Users mailing list? This breaks the reply to sender only / reply to all / list reply functionality of my mailer. It's really broken :( Some would say your mail client is broken. What you're complaining about is generally called 'reply-to munging', and there's been a long discussion about this. Google reveals more, like these two oppositional opinions http://www.unicom.com/pw/reply-to-harmful.html http://www.metasystema.net/essays/reply-to.mhtml Regards, Bruno. And there probably will NEVER ba an agreement on this subject. Another list I am on even went so far as to take a poll, and it was split right down the middle, half taking the correct position outlined in the first article, and half the second, much less flexible, position.. The really curious thing on this list is every so often, if I choose to reply, the poster AND the list appear, but mostly just the list, as if the poster had some control as well. Well, the reason for the latter apparently is that, in some postings to this list, there's actually two entries in the reply-to header, the posters mail and the list address, while in others it's only the list. Why this happens is above me, though, I thought it should be either/or. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep [EMAIL PROTECTED] writes: Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? Well, the group receives it's content by email. It's nothing else than a subscribed user. As that, it could post (email) to this list as well. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep [EMAIL PROTECTED] writes: Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? Well, the group receives it's content by email. It's nothing else than a subscribed user. As that, it could post (email) to this list as well. Regards, Bruno. And that is where the problems starts, if the group posts via email, and is subscribed via email, you form a loop Someone posts to google, google emails the list, the list emails google, google emails the list... Am I missing something simple here? I'd think so, at least from my perspective. I.e. you assume that there's an independent 'posting to google', in which case of course trouble will happen, be it only the duplicates. But as said, as long as the group takes the list as it's *only* feed, and posting to the group is equivalent to just forwarding email to the list, where the envelope sender is the user under which the google group is subscribed and the From is that of the posting user, all should be fine. I don't know at all how it's currently implemented. All I say is that, from the technical pov, proxying any list through such a group should be feasible, without incurring major troubles. Sidenote: I've been thinking myself these days about how to 'mirror' or 'proxy' mailing lists via a news server, with posting capability. The setup detailed above essentially is how I would do it. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reply-To?
Josiah Bryan [EMAIL PROTECTED] writes: On Friday 08 April 2005 1:12 pm, Bruno Hertz wrote: Well, the reason for the latter apparently is that, in some postings to this list, there's actually two entries in the reply-to header, the posters mail and the list address, while in others it's only the list. Why this happens is above me, though, I thought it should be either/or. Though it may be 'technically' correct per RFC guidlines, is it really correct usage-wise? Commen sense tells me that when I click reply, i want to reply to the message, and i want the message to go back to where it came from, in this case the mailing list, not the individual. The individual sent it to the *Mailing List*, not to *Me*. The *Mailing List* then sent it to me, therefore I am replying to the *Maling List*, not the individual. Does that make sense? Yes, RFCs may say different, but are they really logical to the common man? Or even to technical users who dont care about the RFCs and just want to do their work? We're about to get knee deep into that age old discussion I guess :) The point is whether you view the list as a 'sender' or merely as a distribution channel. If the former, all headers should then be rewritten, From, Cc, whatever. And that's definitely not what people want. Now, considering RFC822 and common MUA implementations, current practice is to use * Reply-To, defaulting to From, for a reply * Reply-To, defaulting to From, plus To and Cc, for a 'wide reply' or followup. And here the discussion starts. One party argues that, since the list address is available in the original To or Cc, responding to the list should be done by a followup and eleminating unwanted addresses (i.e. at least the original sender, if Reply-To was untouched by the mailing list). So there'd be always editing involved, and there's a (high) chance of people getting two copies of the same mail by accident. Others say, Reply-To should be rewritten to point to the list, in which a simple reply would go exactly there. Which means, though, that the original Reply-To, which might have differed from the 'From', is lost. Not too good either. Then, people invented the Mail-Followup-To header, which is not standard but honored by some MUAs these days, to store the list address there, and a followup or wide reply should honor this header first before defaulting to the above behavior. But as said this isn't standard and argued about, either. If you look at my mail headers though, you'll see that one included. Either way, what I observed in the other mail, i.e. that some mails on this list have more than one address in the Reply-To header, can afaik definitely be considered broken behavior and should be fixed. This is most likely a mailman misconfiguration. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
tim panton [EMAIL PROTECTED] writes: On 8 Apr 2005, at 20:02, Bruno Hertz wrote: I don't know at all how it's currently implemented. All I say is that, from the technical pov, proxying any list through such a group should be feasible, without incurring major troubles. Given that Google are describing this as a 'beta' perhaps someone who understands this issue should make an RFE to them? I guess rather not. Looking at it a second time, especially at the group description http://groups-beta.google.com/group/Asterisk-test/about one can guess what's happened here, and that most likely neither is google involved nor is this group an actual mirror. My hypothesis: someone created this group, as any registered user can, and just subscribed that group's email address [EMAIL PROTECTED] to the list. Espcially, postings to the group then will not be propagated to this list but stay local there. Most likely just some user creating kind of his own searchable archive while possibly misleading new users to think it might be a legitimate asterisk group. Maybe even a reaction to the recent discussion about forums and stuff. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep [EMAIL PROTECTED] writes: On Fri, 2005-04-08 at 12:01 -0600, Damon Estep wrote: Why? I'd say it's only a config issue. As long as the google group has this mailing list as it's only feed and posting to the group is equivalent to posting to the list everything should be fine. How do you propose getting posts from google to here? Email? Well, the group receives it's content by email. It's nothing else than a subscribed user. As that, it could post (email) to this list as well. Regards, Bruno. And that is where the problems starts, if the group posts via email, and is subscribed via email, you form a loop *ONLY* if you redirect everything google receives via email back to the list. They do not have to do that they could forward only what is posted via their webpage to the list, but choose not to do (aparently) which causes a seperate list populated in part by the existing list. It creates a one way information flow to google groups but not from it. -- Trixter http://www.0xdecafbad.com Do we know who set the group up? Is that an option? No, apparently not. What I was dwelling on is how making a (news)group a read/write list mirror could be done. But as of know, and as I understand it, while you can create arbitrary groups on google and even subscribe them to mailing lists, google provides no means for the other direction. I.e. anything posted to your group will stay there and not propagate. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Roman Volf [EMAIL PROTECTED] writes: I have noticed that many threads don't go as well as planned and wind up in the wrong place. But you do realize that that's not google's fault :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound quality with Xten Xlite softphones...
Maik Hassel [EMAIL PROTECTED] writes: Hello everybody, I started using the XTen Xlite softphones (just to get something up and running quickly). Everything works fine now, but the sound quality is somewhat disappointing. The sending - e.g. everything I say, dtmf tones, etc - receives the person on the other end in perfect quality, everything great. The problem is in everything that I hear: May it be the recorded voicemail message, echo test, or someone talking on the other line, the quality is mediocre at best. It actually sounds like someone is interrupting the line for microseconds (like someone is wiggeling the headphone cable in the plug). Now packet loss is unlikely, I am using other voip product for international calls (like skype) and never encountered that problem, and the test network only consists of two machines and a switch. Did anyone else encounter the problem? Is it a XLite problem (if yes: Can anyone suggest a better free(!) softphone)? Or is it Asterisk related? Most likely an XLite/local sound (driver etc) related problem. I'm currently testing XLite on linux, and especially the XLite buffer parameters have quite some impact on audio quality as well as latency, in a tradeoff sense. You could try other softphones, like SJPhone, and compare. If there's a notable difference, try to tune XLite in case you want to use it, or just use the phone which works best. Look here for a extensive list of hard- and softphones http://www.voip-info.org/wiki-VOIP+Phones Also, it's always beneficial to tell us the OS you're working and having problems on (Windows, Linux, etc), since e.g. XLite on linux is still beta. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
tim panton [EMAIL PROTECTED] writes: On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote: Hi, QoS is nice (and important) but only works within a FULLY controlled end to end link. Inside a BIG enterprise LAN, on leased lines its OK. Using end to end MPLS should also be ok Mind that some provider sell MPLS but it is not their own MPLS end to end. Going from one provider on MPLS to another on MPLS, you lose all the benefits. No control. Using the World Wide Wait (Internet) it will not help. A waste of money. My 2 cents. I'm not sure I totally agree. It is also useful if you control the narrowest pipe. Take the example of several sub-offices joined to a head office PBX over 'public' ADSL lines. Let's say the company buys all the ADSL lines from the same provider. In such a set-up, the uplink side of the sub-office ADSL links are likely to be the main bandwidth limit. A well configured router there will slow outgoing email etc to preserve the quality of current VOIP sessions. Sure, the provider may have internal bandwidth constrictions, but they are unlikely to kick in before the 256k up channel of a typical ADSL. Oh, and, the web and the internet are not the same thing. Think like that and you'll forget mail. Which is a huge bandwidth consumer, and can stand being delayed by a second or two. Tim. I agree, especially qos on upstream might be beneficial, and surely is in a cable modem setup. E.g. my modem has a 10 Mbit LAN interface, but uplink is limited to 256Kbit. So when I have many things going out, uplink will be much sooner saturated than the LAN link, and cable modem buffers run full leading to looong latencies and maybe even package loss. Putting a router before the modem shaping the upstream traffic solves that problem. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bandwidth
Bernie [EMAIL PROTECTED] writes: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. B William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. It can be reduced. Just goole for 'asterisk codecs bandwidth' and click the top link. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email
Francesco Peeters [EMAIL PROTECTED] writes: On the other hand imagine a forum with subtopics like sipura, softphones, zap or whatever. Wouldn't that maybe help to put some load off at least the casual reader and poster seeking or giving advice for topics he/she specialized in, and maybe even the more active participants? Just a thought, and not a bad one imho. Nah, like I said, IMHO it's not different from multiple maillists, as long as the same rules are applied consistenly... ;-) Well, it's easy to say nah if you don't want to think about it. Again, I favor mailing lists too, and all would be OK for me if ppl here weren't already complaining about volume and stuff. So, let me point out two obvious differences you missed: (1) Subscription With a web forum, you register once to the whole forum and have thus access to all topics. On the other hand, when you have like twenty mailing lists on various * topics, who (especially of them newcomers) would subscribe to them all? E.g. if you only have one or two questions to post you'll subscribe to the most introductory/general list and are very likely to stay there. (2) Topic choice With a web forum, you have all topics generally visible on the main page and are likely to see them any time you visit the forum, while when subscribing to lists you do it once and stay. How often do you actually look what other lists are actually available for particular topics? Only if you're forced to, I gather, e.g. because you don't get help on your current list(s). So with mailing lists, there's just higher gravity which lets ppl stick e.g. to -users. Anyway, before saying nah, please keep in mind that I'm not advocating anything right now but just suggesting to keep an open mind since there actually *are* problems with this list ppl have been complaining about for some time. As nothing seems to improve in the current setup, it wouldn't hurt, while discussing this, to at least seriously consider and thoroughly evaluate alternatives. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Are there online forums instead of this
Tim Bass [EMAIL PROTECTED] writes: the excellent movie Vanilla Sky)... Ahem. . . . B#2. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Are there online forums instead of this email forum??
[EMAIL PROTECTED] (Tony Mountifield) writes: I totally agree. I run a local INN server and all the mailing lists I subscribe to get turned locally into newsgroup postings in moderated groups. When I make a posting, it gets mailed out through a filter to the moderator address, which is just the list posting address. Makes handling threads a breeze. Now this sounds like a nice solution, and seems to be one step away from a complete news/mailing list gateway (registration). Did you set this all up yourself? Since I was about to investigate this stuff myself today, i.e. to gateway the list with a standalone news server and then maybe even add a decent web interface with search capablities. I suspect there'll be few 'solutions' out there, since if so you'd run across them more often, but in case you have any pointers I'd sure appreciate them (man, I really like the idea proxying all that lists though inn ... :) ) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email
Francesco Peeters [EMAIL PROTECTED] writes: I think you took my Nah a itsy bit out of context there... ;-) Hehe, I guess context is what your neurons link to - which, as you look at them, might account for the itsyness :) Totally OT: I have been looking at this as a plugin for my own (non tech) WebBBS/Forum, but the problem is that not all clients adhered to the 'references' SMTP-header behavior at that time... AFAIK your observation about broken clients (or broken setups of clients, for that matter) still applies, and makes a strict mail thread - board topic mapping pretty much infeasible. If you abandon that requirement though, a web interface could still be useful, just to interface the lists themselves with reading/posting functionality and searchability. I'll be doing a little search though about this stuff today, maybe something useful comes up ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
Dana Olson [EMAIL PROTECTED] writes: I'm pretty sure that I used SJphone to check my VM. I'll test again. But there is a new beta out on their site (and it's newer than the Windows build). Maybe they added a dialpad? Thanks, Dana, I know keypad dtmf worked with sjphone at some stage, but at the time of my last softphone evaluation roundup some three months ago it was broken. As you know, one doesn't check them all every day, which invalidates statements about many of those linux ports pretty soon as they are apparently still under development. I'll be looking at their last build soon, though, and if only the keypad behavior was fixed it would, as said, imo make sjphone a viable alternative. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Are there online forums instead of this email forum??
[EMAIL PROTECTED] (Tony Mountifield) writes: Yes, based on a standard install of the INN rpm in Red Hat or Fedora. I've just put together a page with a description and links to the two perl scripts used. See http://www.softins.co.uk/mail2news Geez, right on time :) I just installed inn and was thinking about how to glue it to mail. From what I learned through google, the whole matter is not entirely trivial, so your effort is most welcome and highly appreciated. My mail setup differs slightly (postfix/cyrus, no procmail), so I'm not entirely sure yet were to plug the mail-news feed, especially since I don't want to do user specific filtering on the postfix side. Maybe via cyrus/sieve ... Those are minor issues though, apparently you got the ground pretty much covered, so many thanks for that! Incidentally, did you also already think about what it would need to make such a server public, including posting? As I'm writing I'm beginning to think this might even be not possible for various reasons, e.g. even if one got news auth and list subscription synced, users would still get the mail, too ... seems to need a pretty tight coupling between maling list and news server. Hmmm ... anyway, we'll see, one step after the other :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
hank smith [EMAIL PROTECTED] writes: do you know if it is gtk2? It appears to be: $ ldd xlite-linux-22 ... blah ... libgtk-x11-2.0.so.0 = /usr/lib/libgtk-x11-2.0.so.0 ... blah ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
Kris Edwards [EMAIL PROTECTED] writes: Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends from my disgust with everything else. In particular, kphone, and sjphone. I have noticed latency with xten in meetme, but if I just dial somebody it works better than anything I've tried (so far.. I've only spend about 1 hour talktime). Anyway, I'm certainly more hip on open source, and can't wait to try gnomemeetings sip once I can actually get it to compile :/ I have not tried lipz4 yet either (not sure if it will work w/ gentoo, but I might give it a try if I can find any rave reviews) Funny how experiences vary. E.g. I thought sjphone for linux wasn't too bad, if it only had a dial pad. Anyways, I'm trying the latest xlite beta right now, and I must say it really has improved. I've been sticking to gnomemeeting yet, but here seems to be a candidate to be taken into serious consideration for everyday use. Especially since the GM/SIP support apparently takes it's time. Regarding my previous statements, echo tests with a local and a public * server are pretty fine now, and audio/latency is way better compared to my last tests maybe two months ago. In this respect, the current beta actually does equally well as GM and could be considered fit for production. What I personally don't like though is that funky interface where one even can't always be sure where the mouseclick 'hot spots' are. For an mp3 player this might be OK, but this being a tool one is supposed to really work with I'm not sure what's going through the mind of those people. Hopefully, a reasonable skin comes up some time in the near future ... Still, thanks (finally) for reminding me of a phone that I've put aside maybe a little too early :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues
Henry Devito [EMAIL PROTECTED] writes: Forget this post I had a typo in my voicemail.conf file sendvoicemail=yes was spelled wrong. That fixes point 1) What about the others? - Original Message - From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 31, 2005 9:04 AM Subject: [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues Folks! I want to let everyone know that I have been trying to migrate from 1.0.6 to 1.0.7 last few days and I have come across serious issues in the build 1.0.7. What I found are listed below. I would recommend everyone to hold off any upgrade till the next build. 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say extension numbers for the called user. My guess is all these .gsm files are corrupt and hence you don't hear anything. 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are corrupt. When we tried to play these files using a media player, all we hear is gibberish. 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we configure this for RFC2833. Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been able to find a fix? Seshu Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
Dana Olson [EMAIL PROTECTED] writes: I've been meaning to try it again. A large number of builds have been sent since I last tried. And boy, it was sooo slow and more resource-intensive than its Windows counterpart. Maybe, but I still recommend trying again. It's really making headway. I haven't been using a softphone at home because I'm waiting for GnomeMeeting w/SIP to get into Ubuntu or Debian. I don't expect them to come up with anything usable before another 6 months. Actually, I'd be surprised if it happened this year. Instead, I just use a cordless phone plugged into my TDM card. Well, that's even better than any softphone, isn't it? Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email forum??
Andrew Kohlsmith [EMAIL PROTECTED] writes: Call it archaic if you like but mailing lists get the job done faster, better and without all the bullshit that forums bring to the table. It's not archaic but reasonable. Clicking around in a funky web interface is definitely not what I call productive communication when compared to what good email clients (like gnus :) ) can do for you. My order of preference would be news groups, mailing lists, then everything else except web forums, which comes last. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
Dana Olson [EMAIL PROTECTED] writes: What's wrong with using your keyboard's Num pad? Nothing. Tried that, didn't work. Build 1.30.256b ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email
Martijn van Oosterhout [EMAIL PROTECTED] writes: Ok, basic use case. I today go to a forum and read all the messages. Next day I come along, how do I get a list of all the messages I havn't read in thread order in such a way that if I decide to go somewhere in the meantime, it knows what I've read and what I havn't. I also monitor several other projects all on mailing lists. With one mail box I can monitor six projects in one interface. I don't touch the mouse the whole time. I can whizz through a message every few seconds because every one is in the same font, same colour, same spacing (HTML all disabled). No forum is ever going to compete with that sorry. This really is a killer argument, and I wholeheartedly agree with that. One point comes to mind though, which has been troubling people here for some time and where web forums, as much as i dislike them, could actually be of use, i.e. partitioning. As of now, all kinds of stuff is thrown into this list, mostly * related but not always, from whatever cards over sipura products and manager api to softphone setup and whatever. Now, even if mailing lists were set up for particular topics, I think experience tells us that quite a few users would come here anyway, and people would have a tough time educating them. That's I think the main reason no serious effort is taken in that direction. On the other hand imagine a forum with subtopics like sipura, softphones, zap or whatever. Wouldn't that maybe help to put some load off at least the casual reader and poster seeking or giving advice for topics he/she specialized in, and maybe even the more active participants? Just a thought, and not a bad one imho. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
Brian Capouch [EMAIL PROTECTED] writes: Hmmm. I just got the latest beta build, which identifies itself as 1105d. The keypad functionality is perfect. Hmmm. Good for you. We were talking about sjphone, though :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
Kris Edwards [EMAIL PROTECTED] writes: This is the best linux sip phone I've used so far. Audio quality has been perfect and it seems really stable, so hopefully it will be out of beta soon. I might actually pay for the full version! (not counting console games, that would be the second piece of software I've purchaced since 1987). Sounds rather like you want to sell the full version. Myself, I don't know about recent betas since, frankly, I didn't care anymore after initial experiences being pretty much disappointing. The first beta I got produced no audio at all, and we had a tough time to convince the developer that it wasn't a driver issue. The next releases then had huge latencies, primarily due to the Xlite audio setup. Now, I admit that setting up audio for interactive/'realtime' apps on linux is a mess, but various open source projects have already done much better. So no, in contrast to your plug I'm not as enthusiastic myself, especially since audio quality resp. latency is the one major trouble I had with linux softphones. E.g. iaxcomm would be great and totally satisfying for me if latency were (significantly) less than 1 second. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging Asterisk in GDB (DDD)
Jay Ray [EMAIL PROTECTED] writes: Thx manI will try to start it from withing DDDNo one responded in DEV list No one answered because your question was way too dumb (sorry). If you attach with a debugger to a running process, the process will be stopped. You then have control of it (step forward, run, etc) in the debugger itself. This is very elementary knowledge one can expect even from a newbie junior freshman developer. Don't expect * users resp. developers to fill you in on topics totally unrelated to asterisk. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gnomemeeting / sip
Kris Edwards [EMAIL PROTECTED] writes: I've seen some posts about ppl using gnomemeeting via oh323, but is anyone using it w/ sip?? (only their cvs supports sip, but I figured somebody was trying it.. I'm grabbing it now :) I tried to get GM/Opal going some six weeks ago but it didn't even compile. When asking, Craig/Damian advised to better wait a little. Considering that Craig was on vacation the last month I'd be surprised if things have noticably improved in the meantime. Still, if you get it going, a short writedown on the steps you took would sure be appreciated :) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring GnomeMeeting for Asterisk
On Fri, 2005-03-18 at 20:33 +0100, Stefan Stolz wrote: Hello, i tried to configure Gnomemeeting for Asterisk, because its, how it looks, the only tool which gifes me all i want for the use in linux... I have allready installed and running h323 support in asterisk and edited the h323.conf. But i have no chance to configure Gnomemeeting that it connects with Asterisk! I found also nothing useful in the web Can anyone tell me what settings in GnomeMeeting i must take, that Asterisk can lead calls to GnomeMeeting and GnomeMeeting leads calls to Asterisk? It looks like that the settings for Gatekeeper arent right :-( Thanks! - -- Grsse Stolzi If you run * without gatekeeper (e.g. gnugk), disable it in h323.conf and configure gnomemeeting to use * as a proxy (Gateway/Proxy Settings). Then, for calling out, specify an appropriate (default) context in h323.conf. To receive calls with GM, you have to add a line like exten = yourexten,1,Dial(OH323/yourip:1720) to the context which handles your incoming calls. 'yourexten' here is the extension under which your client should be reachable (which can be a number or the 's' default), and 'yourip' the IP address or domain name of the machine on which the GM client is running. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring GnomeMeeting for Asterisk
On Fri, 2005-03-18 at 22:02 +0100, Bruno Hertz wrote: To receive calls with GM, you have to add a line like exten = yourexten,1,Dial(OH323/yourip:1720) to the context which handles your incoming calls. Correction: exten = yourexten,1,Dial(H323/yourip:1720) It's because I use OH323 and not H323. Sorry. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Basical question to asterisk
On Wed, 2005-03-16 at 13:13 +0100, Christian Schoepplein wrote: Hello! I'm new to asterisk and because I try to configure the package for my needs the last days without success, I'd like to ask a basical qestion. I need asterisk to work together with the German VoIP provider sipgate (http://www.sipgate.de). Asterisk should act as a softphone, I want to recive and make calls only with the software under linux, no softphone should be used. Is this possible with asterisk in principle or do I have to use a real softphone together with asterisk? Manny thanks! You can use asterisk as a softphone with either chan_oss or chan_alsa. Googling for 'asterisk' and 'softphone' gives this link at 7th position http://www.junghanns.net/asterisk/page13.html It's slightly outdated, you won't need the diff any more (as far as I can tell), but it still gives you the general idea. *'s softphone capabilities are somehow limited though. E.g. you can't put calls on hold, and what bothers me even more is that the soundcard isn't released between calls. I.e. * grabs it on startup and releases it only when quitting, unlike (most) other softphones. On the other hand, latency wise * is the best softphone I came across on Linux. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to connect with a headphone
On Tue, 2005-03-15 at 20:09 +0100, Andreas Meyer wrote: Sorry for not being clear enough but my headphone is attached to the soundcard at my local PC. Now when I start Asterisk on that machine it is using port 5060 and sjphone can not connect because it also uses port 5060. netstat -panu |grep 506 udp0 0 0.0.0.0:5060 0.0.0.0:* 5773/asterisk If I could tell Asterisk to listen to more ports than 5060 it would be no problem. OK, that really wasn't clear from your first posting. The problem is that you try to run two SIP 'clients' on the same machine, sjphone and asterisk, and both try to bind to port 5060 for that reason. It has literally nothing to do with your headset/phone. I you want asterisk to take SIP calls, you can't really run sjphone on the same machine. * itself however can act like a softphone via chan_oss or chan_alsa, so you might want to look into this (* wiki) and just drop sjphone. If asterisk doesn't need to take SIP calls, just disable chan_sip loading and sjphone should have no problems running. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip-Info
On Tue, 2005-03-15 at 16:05 -0700, Zanzamar Majere wrote: Is anyone else having issues pulling up voip-info.org? There's been a 'wiki down' thread running all day on this list. So it's been noticed, yes. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Am i right by Asterisk?
On Fri, 2005-03-11 at 15:32 +0100, Stefan Stolz wrote: Hello, i tryed to read the Wiki, but i am not sure if i am right with Asterisk. Until now i made my phone calls with ant-phone over my ISDN Fritz Card. Now i tryed to search a way to phone from other computers in the internal net over the Fritz Card on the Server. Someone told me Asterisk can do this. I read in the Wiki that Asterisk is in special for Voip, but it looked like that it can also make ISDN calls. Can Asterisk do this? What do i need to phone with Asterisk over ISDN into the phone net? Or where can i read about this things? I think all i need stands in the Wiki, but it was to much for me to find the right thing out for me... I think i need ISDN4Linux, because ant-phone used this and it worked. I read that i need a special Plugin for this because Asterisk per default cant do this? Can you help me to get order in my confusion? ;-) Thank you very much! Stefan to clarify what you want to achieve: you have an Fritz ISDN card and what to issue calls from several computers on your LAN to the ISDN line, right? If that's your question, the answer is yes, asterisk can do this, and I have exactly that setup LAN | | Host1 - |- NAT- Internet Host2 - Asterisk Server Host3 - |- Fritz - ISDN For the communication between your computers and asterisk, you'll use some VoIP protocol, like SIP, IAX or H323 and a corresponding client (SJPhone, Iaxcomm, Gnomemeeting). Regarding asterisk interfacing the Fritz card you might either use chan_modem and isdn4linux, which I didn't test myself but it seems it's not very recommended, or chan_capi and the AVM capi drivers, which I have running myself and work OK. Another alternative is mISDN. Finally, you'll need to setup a proper asterisk dial plan to link all that together. It's not trivial at the beginning, but doing some reading especially on the Wiki and in mailing list archives will help you a lot, so it's not too hard either. Good luck Bruno ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does asterisk do the routing?
On Tue, 2005-03-08 at 17:13 +0100, Michael Vogel wrote: So I want to register the SIP client at the asterisk server that itself is registered at the different SIP providers. Does that work the way I want? It's what people do here all the time. One issue might arise though, i.e. where your * server is, on the internet or on your LAN. If the latter applies, make sure your router allows registering his inbuilt client with LAN servers, too (so it's probably rather about how the router does the routing). In that scenario, of course all the issues discussed here at length, like SIP and NAT troubles, apply too. With your * server on the internet though you should be fine. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2 gui (out of tree modules)
On Fri, 2005-03-04 at 18:17 -0800, Dan Austin wrote: There does not seem to be too much interest in this, but it has helped me sell the idea of dumping a very expensive, but poorly functioning, existing VoIP conferencing system. In the future I can send announcements directly to the few people who expressed an interest, to keep from using list bandwidth if there is no general interest. No, no. Though I'm little likely to investigate into your software in the near future, these kinds of announcements are always very interesting and welcome I'm sure most people here will agree. So you may well keep posting them here. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
On Sun, 2005-02-20 at 13:51 -0500, Paul wrote: Or maybe a double fool because he also disrespected Debian GNU/Linux in his reply. *And* recommended Fedora, which makes it triple. I just dumped FC3 and replaced it with Debian because Fedora's kernels constantly gave me issues, e.g. with proprietary AVM kernel drivers which didn't even work. On the other hand, no probs whatsoever with Debian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
On Tue, 2005-02-15 at 13:59 +, Alistair Cunningham wrote: It can also handle video calls, though I have not used this myself. AFAIK video only with SIP, which I didn't test myself either. With H323 it does not work, audio only there. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linphone / Kphone
On Mon, 2005-02-14 at 22:29 +1100, Duane wrote: I gave up trying to use linux soft clients they all seem to have some fatal flaws or issues I could never fully get rid of While I'd second that, Gnomemeeting is still pretty good and by far the best softphone I've used on Linux. Currently, it supports H323 only, but SIP support is in development. It looks like it will take some more time though until a first test version is available. http://mail.gnome.org/archives/gnomemeeting-list/2004-December/msg00198.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linphone / Kphone
On Mon, 2005-02-14 at 10:47 -0700, Kyle Hagan wrote: I used to use kphone and have very bad echo, I switched to sjphone and it worked great. It isn't too bad, but it has latency (compare it e.g. to asterisk as softphone and you'll see what I mean) and no dial pad. So I found it isn't really satisfying either. Another point to note is that seemingly all closed source softphones (SJ, XLite beta and also cornfed) make connections to web servers and transmit platform/call information. Don't know how you think about that, but for me that's behavior I'd like to avoid if ever possible. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linphone / Kphone
On Mon, 2005-02-14 at 14:22 -0500, Dana Olson wrote: Do you have this documented somewhere? Is this for the Linux Xlite and SJphone only, or the Win32 ones as well? We're talking Linux currently, don't know about Windows. Documented? On the cornfed website it's specifically mentioned, with affirmation that it's not meant to be spyware. XLite Linux is still beta, so one might argue that's currently for debugging, which is OK. It'll be interesting to see whether it's turned off in the production version. SJPhone sent unwanted traffic the last time I checked some months ago, but the version I currently have (SJphoneLNX-256b) seems to be clean. So correction on this point. Generally, running tcpdump or watching your firewall log should tell you what your apps are doing. The data I've seen being sent has always been sufficiently general/anonymous to not feel paranoid about it, so I sure didn't want to raise an alarm. Regard it rather as a matter of taste, if you will. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linphone / Kphone
On Tue, 2005-02-15 at 11:43 +1100, Duane wrote: Yea I've been hanging out for them to support it for ages now... Hehe. Not the worst thing to hang out for :) Anyway, OPAL seems to have a reasonably working SIP stack by now, I did a test run with the cli client and it worked. Some features are missing maybe, didn't check in detail, and GM integration is still in the works but shouldn't be too difficult I guess. So restrained optimism might be in order, at least regarding preliminary test versions during the coming months ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
On Sun, 2005-02-13 at 04:46 +0100, Andres Gmez Garca wrote: I've tried GNOMEMeeting also. It works fine with a P2P client connections (ALSA works fine) but, even when I success connecting to an asterisk server, I haven't hear anything. I mean, I don't hear the demo successfull messages. I've looking the GNOMEMeeting logs and it says that it closes the sound channel as soon as it connects to the asterisk server. This is my h323.conf file: Had the same issue with Debian Sarge. I didn't actually investigate it, but I strongly suspect the openh323/pwlib packages don't work with the asterisk-h323 package. The H323 README specifically says btw to don't use the packages of the distribution but rather the versions recommended there. I finally decided to compile * 1.0.5 from scratch, as well as use chan_oh323 instead of chan_h323, and all works well now. As to the linphone problems, don't know, it should work. If not, it'd be rather a linphone issue. As to an IAX phone, the only choice on linux currently seems to be iaxcomm/iaxclient. For me, it's not really usable because of latency issues, but to test the * installation it'll suffice anyway. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
Addendum: I did a little investigation and found this http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259 Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
On Sun, 2005-02-13 at 18:10 +0100, Andres Gmez Garca wrote: Thanks Bruno, I'll try it. Also, you might take a look again at http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259 Following your mail, I wrote to that list (cf the last mails there), and it looks like a working oh323 package will turn up soon. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Searchable Mailing Lists NooB Question
On Thu, 2005-02-10 at 21:44 -0600, Steven Critchfield wrote: So you probably want to still turn off the webserver and jabber server, they would be better off coloed anyways and there are a lot of cheap colo places for non critical hosting. As a sidenote, you can also set up traffic shaping to prioritize particular traffic/ports. I.e. if it's OK for you to starve web and jabber clients during voip calls, you can still run those servers without impairing your voice streams. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk@home scary log
On Thu, 2005-02-10 at 10:47 -0600, Steven Critchfield wrote: This is a good example of why ease of use is not always a good thing. Had you actually had to learn more before you had an install, you would have been through a text or two that mention password strengths. Apropos ease of use: on publicly accessible servers I disable OpenSSH password access anyway, and allow login only by key. The key passphrase never travels across the net, and per ssh-add it can be stored by an agent which keeps it in memory until log off from your desktop session. I.e. you have to type it only once. Altogether, this gives much more security together with maximum ease of use. Also, just fyi, ssh account and password guessing resp. cracking seems to be hip right now, since I see attempts in my log on a daily basis. Fortunately, with passwords disabled, and as long as OpenSSH itself isn't vulnerable (buffer overflow etc.), I really don't need to be paranoid about this Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk@home scary log
On Thu, 2005-02-10 at 11:09 -0500, Jason Stewart wrote: There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. You sure? I run postfix myself and am not proficient in analyzing sendmail logs, but looking at those lines Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) I find the relay (accepting host) is 127.0.0.1. So, even if ignoring the envelope 'from', there seems to be no doubt which host this mail was sent from. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home scary log
On Thu, 2005-02-10 at 10:42 -0600, Daniel Wright wrote: You can always set up ssh to use host keys. Here are two howto's on what else? How to set them up. http://www.securityfocus.com/infocus/1806 Part 1 http://www.securityfocus.com/infocus/1810 Part 2 Great links. One may add that first actually deals with host keys, which identify the server to the client, and the second with identities resp. pubkeys, which identify the client to the server. I guess it's actually the latter item we are currently talking about. Host keys are essential as well of course, to prevent phishing and the likes. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk@home scary log
On Thu, 2005-02-10 at 09:57 -0700, Colin Anderson wrote: 5. Use key-based auth mechanism rather than password. It's my understanding that the key is never sent, only a hash of the key. The target system compares the hash against it's hash of the key, and if it matches, cool. Not exactly, for the sake of completeness. Public/private key authentication usually is based on the fact that messages encrypted by a public key can only be decrypted by the private key. So your public key, which is stored on the server, can be used by the server to send an encrypted challenge. If you are able to decrypt that challenge, via the private key stored on your desktop system, you've proven that you have the private key and hence are the identity you said you are. So, whoever has access to the private key, and to the (optional but vital!) passphrase with which the key is encrypted for storage, can authorize against the corresponding public key. That's why the private key and it's passphrase must be kept secret. On the other hand, all that travels the net are arbitrary one time challenges, and no critical information is exposed. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /dev/dsp blocked
On Thu, 2005-02-10 at 13:14 -0400, [=Jorge Boscan Etura=] wrote: Hi I just installed * 1.0.5 on a my fc2, but when i try to use kphone to begin testing it doesnt work because * is using /dev/dsp, how can I configure/resolve this? The * modules which may use the sound card (and enable you to use * like a softphone) are chan_oss resp. chan_alsa. Disable them in modules.conf via noload and you should be fine. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home scary log
On Thu, 2005-02-10 at 13:30 -0600, Steven Critchfield wrote: Nothing like sending a valid key to a man-in-the-middle. There is no key sent to man-in-the-middle except the pub one, which does no harm anyway. What does harm is you're logging into/routing through a different machine that you think, which then simply spies on your actions, like su or whatever you do. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More complicated huntgroups / delayed ringing - SOLVED
On Tue, 2005-02-08 at 22:50 +0100, Stefan Gofferje wrote: Queues are not too bad but lacking an important feature. As far as I could figure out, they couldn't just ring without answering. Hum? I'm running * 1.0.5, and Queue rings without prior answering the line. Although most queue examples seem to assume you want MOH and stuff, which of course implies a prior Answer, it isn't actually necessary. I was specifically concerned about this point, since in my home setup I receive ISDN calls via chan_capi, and while directing them into the queue I only want the caller to pay if either a phone or voicemail picks the call up. Maybe I missed something, as I didn't follow the entire thread, but Queue ringing without Answer works here. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More complicated huntgroups / delayed ringing - SOLVED
On Wed, 2005-02-09 at 00:37 +0100, Remco Barende wrote:
Could you post your configs please? I need the same for my home setup :)
Sure, but there's nothing sophisticated:
extensions.conf
---
[myqueue-in]
exten = s,1,Queue(myqueue25)
exten = s,2,Macro(vm,${MYEXTEN})
exten = s,3,Hangup
[default]
exten = ${MYEXTEN},1,GoTo(myqueue-in,s,1)
[macro-vm]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,VoiceMail(u${ARG1})
exten = s,4,Hangup
queue.conf
--
[myqueue]
strategy = ringall
timeout = 10
retry = 5
member = SIP/10 ; my registered SIP softphone
member = IAX2/10/[EMAIL PROTECTED] ; my registered IAX softphone
member = OH323/caruso.quasi.local:1720 ; my H323 softphone
member = Console/dsp ; at home this is OK I guess
That's all, rather primitive actually (you might want to tune timeout
and retry vars in queue.conf though to give you a proper timeout before
VM picks up).
Regards, Bruno.
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Re: [Asterisk-Users] Help with chan_h323
On Fri, 2005-02-04 at 15:54 +0900, Andrew Kochetkoff wrote: EndedByTransportFail You should give more details, all there can be seen from your log is that the call is dropped due to EndedByTransportFail, which can have various reasons. Apparently, this is a LAN call with no NAT involved, right? But what client (software) is involved in the call, how does your oh323 setup look like? What are your asterisk, chan_oh323, openh323 versions? Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Disabling native bridging for IAX calls
On Wed, 2005-02-02 at 09:09 -0500, Nabeel Jafferali wrote: notransfer=yes That prevents transfers but not bridging. As to my knowledge, there's no way to prevent bridging. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Disabling native bridging for IAX calls
On Wed, 2005-02-02 at 14:36 +, Gareth Blades wrote: If that is the case then it seems a serious limitation as it makes call parking and attended transfers unusable. Your only choice is to use the IAX native transfer where you cannot speak to the recipient before transfering the call. OK, first let's clarify what * we're talking about, which is stable 1.0.5 in my case (especially there's no attended transfer feature). As to the implications of unpreventable bridging I'm not sure. All I know is that with notransfer=no * first tries bridging and then transfer, where the latter usually fails in my case for whatever reasons (NAT? iaxclient?) So I disabled transfer per notransfer=yes for all my peers, and in this case * still tries and succeeds in bridging when channel codecs are the same. Apparently, this then prevents * from interpreting dtmf signals, especially # for blind transfer. Finally, to my knowledge and after (the usual) thorough doc, web and archives searches, there seems to be no way to prevent bridging. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Asterisk to Find a Live Person
On Wed, 2005-02-02 at 14:21 -0800, Aaron Glenn wrote: That's the least ambiguous subject I could muster. I'm relatively new to Asterisk and while I'm certain there is a way to do this, I'm unsure how. My question is this: How do I take an incoming call, put the person on hold, and in the background (i.e. while they are on hold) begin trying other phone numbers until someone answers? The feature you are looking for is call queues: http://www.voip-info.org/wiki-Asterisk+call+queues Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323
On Tue, 2005-02-01 at 14:09 +0900, Kuniyoshi Murata wrote: Hi, I'm thinking of setting up Asterisk for H.323 video phone clients. Now, what is the difference between native H.323 that come with Asterisk and Open H.323 for Asterisk ? I can't tell you the exact differences, but oh323 seems more stable to me, and more actively maintained also. It looks like the native h323 support doesn't have such a high priority amongst * developers. I've been using oh323 for maybe two months now in my home setup, and it works OK. With native h323 I even had troubles get it compiling. Let me point out though that neither h323 nor oh323 support video. The only channel driver I know about which supports video is SIP, and I can't comment on the quality since I didn't test that myself. But if you need video with h323, asterisk won't work for you. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stableasterisk
On Tue, 2005-02-01 at 13:21 +0100, Robert Rozman wrote: By the way: use asterisk-oh-0.7.x! But shouldn't I use 0.6.5 cause I'm on cvs STABLE ? You are completely right. 0.6.5 for STABLE and 0.7.x for HEAD. Regards, Bruno, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX native transfers
On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote: Unattended transfers just does nothing. I cannot get it to do anything. Not sure about this, but I'm under the impression that the # transfer might need some client support. E.g. I tried gnomemeeting - * - NAT - * - firefly and # did nothing. But when using sjphone instead of firefly it worked. So my guess is that when sending the callee to a different extension, the callee's client must support it. Or it may actually be an IAX problem, as sjphone is SIP of course. Didn't try another IAX client, so a definitive answer would interest me as well ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX native transfers
On Tue, 2005-02-01 at 12:43 -0600, Eric Wieling wrote: # transfers are controlled by features.conf and the t and T option on the Dial line. It requires NO support in the client. In fact # transfers are usually only useful if the client does not support NATIVE TRANSFERS, i.e. real non-asterisk handled transfers. Thanks for clarifying this. Any idea then why # transfer enabled via Dial T works with sjphone as callee client and not with firefly? Is it due to IAX? Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX native transfers
On Tue, 2005-02-01 at 17:49 -0200, Denis Galvo - iSolve wrote: I believe that your problem is related to DTMF problems with your softphones. Rather not, since the caller client was gnomemeeting in both cases, and the caller tried to transfer the callee in both cases. With sjphone as callee it worked, with firefly not. Additionally to that, I enabled debugging mode on the * servers, and could see that dtmf # arrived from gnomemeeting properly. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX native transfers
On Tue, 2005-02-01 at 21:29 +0100, Philipp von Klitzing wrote: Gnomemeet -- * -- * -- Gnomemeet? Gnomemeet -- * -- * -- firefly resp. sjphone, as said. and insert codecs and protocols everywhere. alaw gsmbridged for firefly Gnomemeet --- * --- * --- firefly/sjphone, gsm for sjphone H323 IAXIAX firefly Gnomemeet --- * --- * --- firefly/sjphone, SIP sjphone I've seen these problems particularly when * -- * over IAX with GSM was invovled. And of course state your Asterisk version. And then there is the question which * should interpret the #, is it *1 or *2? OK, that'll be the answer. Unfortunately, there's no way to prevent native bridging. So I guess because of that the # isn't interpreted by the second server, as it is in the case of sjphone, although * technically still is in the media path. Right? Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft phones that _actually_ work under Linux?
On Wed, 2005-02-02 at 07:12 +1100, Howard Lowndes wrote: Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - lowww iaxcomm - needs some strange widgets various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with adequate performance with FC2 or FC3. I don't mind whether its SIP or IAX2 - I just need it to _work_. Sounds familiar. For me (FC3) the best option is gnomemeeting, which is still H323 only though (SIP support is underway). I have it working with */oh323, and the whole setup is very satisfying, i.e. robust, low latency, good sound quality. Still, if you need to register things might get messy, i.e. involving a gatekeeper and stuff. Since I have an * 'proxy' in my home LAN I don't need that stuff, cause I can set up the dial plan as needed. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Asymmetric Latency
On Sat, 2005-01-29 at 15:48 +0100, Zdik Kudrle wrote: I'm running Asterisk with HFC-S card connected to HW PBX in my office. When I make a call from home using iaxComm connected to Office Asterisk, the outgoing latency is about 0.25 sec, which is quite OK. But to incoming latency begins on 0.5 sec and in a minute it's about 5 seconds (!) and growing fast. Just curious: how did you measure that latency? Since I have constant trouble especially with iaxcomm - * configurations myself. E.g. currently I have a situation iaxcomm - * - NAT - public * with latency issues. If on the other hand I use gnomemeeting instead of iaxcomm, all works well. The problem for me is to differentiate between soundcard and network latencies. I.e. I think that iaxcomm sets up my soundcard pretty badly, and after digging the source code I found that setting PA_MIN_LATENCY_MSEC=10 as environment variable improves things quite a bit. There are still latency issues though, which I thought I could pin down to transfer attempts, but I'm now bound to dump that hypothesis since notransfer=yes an all servers doesn't change a bit. Finally, debugging this stuff, and especially the network traffic and protocol interaction, is pretty tedious, so if you could give a hint how to best measure network latencies I'd sure appreciate it. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoiding queue retries without hangs?
On Thu, 2005-01-27 at 16:14 -0600, Eric Wieling wrote: You might consider upgrading to 1.0.5 release Thanks, I checked it out. With same config as for 1.0 I get: Asterisk Ready. -- Accepting AUTHENTICATED call from 192.168.0.10, requested format = 1024, actual format = 1024 -- Executing Goto(IAX2/[EMAIL PROTECTED]/2, gh-queuein|s|1) in new stack -- Goto (gh-queuein,s,1) -- Executing Queue(IAX2/[EMAIL PROTECTED]/2, ghq20) in new stack Floating point exception Not exactly an improvement. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoiding queue retries without hangs?
On Thu, 2005-01-27 at 20:35 +0100, Bruno Hertz wrote:
Anybody found a way around this (bug?), i.e. avoiding retries with
Queue(...|t) properly timing out at the same time ?
OK, I took a look at app_queue.c, and while the described behavior isn't
a bug, I still hacked the source to give me a different retry semantics.
Specifically, if retry=0 the original strategy is to set it to a default
value of 5. My hack is to don't do any retries in this case anyway and
behave the same way as if the call timed out on the queue.
For those interested, the changes to app_queue.c are small:
in reload_queues
- if (q-retry 1)
+ if ( (q-retry 1) (q-retry != 0) )
q-retry = DEFAULT_RETRY;
in queue_exec
/* Leave if we have exceeded our queuetimeout */
if (qe.queuetimeout ( (time(NULL) - qe.start) = qe.queuetimeout) ) {
res = 0;
break;
}
+ if ( (qe.parent)-retry == 0 ) {
+ res = 0;
+ break;
+ }
That's it. That way, no retries are attempted at all if retry=0, and
Queue times out if it does so on the queue members, i.e. according to
timeout in queue.conf. Tested though only with ringall, don't sure how
it works with other ringing strategies.
Thanks, Bruno.
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[Asterisk-Users] Avoiding queue retries without hangs?
Talking * 1.0.12 here. Problem: I'd like to avoid retries with queue, i.e. if members choose to ignore a call they should not be bothered again. On the other hand, when a call times out according to the Queue(...) timeout, the call should proceed to voicemail. Setting retry in queue.conf to a high value unfortunately doesn't solve the problem. More specifically, the timeout t given to Queue(...|t) doesn't govern the retry value, and especially when retry t incoming calls just hang in case nobody picks up, and do not proceed to voicemail. Especially, t does not set a hard limit for in queue stay time. Anybody found a way around this (bug?), i.e. avoiding retries with Queue(...|t) properly timing out at the same time ? Thanks, Bruno. PS: fyi, Qeue option 'n' has no impact on the above situation. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoiding queue retries without hangs?
Just a point of order, there is no Asterisk 1.0.12. The latest is 1.0.5. Sure, sorry, it's actually 1.0, i.e. CVS-v1-0-12/18/04-22:40:47. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote: seem like some kind of port issue... Probably. Both try to set up listeners on the IAX port (4569 for IAX2). Disable or reconfigure one of them to bind to a different port, whichever you want to answer on it. Also, don't forget to disable chan_alsa and chan_oss in modules.conf. When running another client you won't want the * console hogging your soundcard. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anybody a patch for oss/alsa to not constantly hog the sound card?
The subject says it all. After digging through latency and other issues with all kinds of linux softphones, I've found that only * works alright for me as a VoIP client. Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab the card once and won't release it until shutdown, while other clients are friendly enough to grab the card only on calls. So, before getting lost in a regular coding frenzy, there isn't by chance any of you who already patched either of those chans to behave a little more cooperative? Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headset with X-Lite
On Thu, 2005-01-20 at 14:51 -0800, Manjit Riat wrote: Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is there any way to configure X-lite to use the headset instead of the speakers? Or will I have to plug the headset in the speaker jack ? Manjit a delicate question, but are you sure that this is an asterisk issue? Because, and I'm confident you won't mind me being frank, this rather sounds like being at most an XLite question, if not only an issue about how to properly connect your headset. Anyway, here's the link to the XLite support forum: http://support.xten.net/ I wish you good luck there. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Headset with X-Lite
On Thu, 2005-01-20 at 16:59 -0800, Manjit Riat wrote: Oh sorry... just got carried away with all the help I got here. No problem. Don't know about your headset, but usually it has two connectors, which you plug into the mic and speaker jacks of the sound card. XLite itself doesn't really care whether you have a headset or not, you could also connect speakers and a microphone. In each case, XLite should access the right channels. You might want to check that your volume settings are in order. If things really don't work, there is an option menu setting in XLite where you can check if it properly recognized the sound card, just don't remember where it was. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with registering Windows Messanger with asterisk
On Tue, 2005-01-18 at 09:31 -0600, Bartosz Wegrzyn - asterisk wrote: I am trying to register windows messanger with asterisk and it fails. http://www.voip-info.org/wiki-Asterisk+phone+Windows+messenger Check whether it's the realm. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Offtopic: improving softphone latency on Linux?
Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly, Messenger is more tightly integrated with the OS and accordingly tuned. So where does this time go? Kernel? Application level? Web searches seem to suggest that sound latency generally is a problem on Linux, so I tried the low latency kernel from http://ccrma.stanford.edu/planetccrma/software/planetccrma.html (there are two kernels, actually, where I only got the stable version to boot - bleeding edge didn't do on my machine). Still, that kernel did not really improve things in a noticeable way. Question hence: did some of you guys experience and investigate this same issue? Any recommendations or hints how to make VoIP even more enjoyable on linux? I wouldn't care that much if I was the only affected party, but of course whomever I call will also suffer from those delays, so as the staunch Linux advocate I've been so far I'd really like to show better performance ... Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?
On Mon, 2005-01-17 at 16:51 -0500, Steve Kann wrote: What softphone are you using on Linux? iaxcomm, linphone and sjphone, and they all give If you use an iaxclient-based softphone on linux as root, it runs with RT priority, and pretty low latency Hmmm, on my side I can't say it makes much of a difference for iaxcomm. It does improve sound quality though, since running iaxcomm non root produces pretty crackling audio, for whatever reasons. Altogether, I find that sjphone performs best, regarding quality as well as latency, where Windows/WM still seems to play in a different league. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?
On Tue, 2005-01-18 at 07:43 +0800, Steve Underwood wrote: Latencies that big should not be due to the softphone. They are often due to the sound card driver. Yeah, it's what I thought, but then, as said, I tried the planetccrma kernel and drivers, which are supposed to support professional audio applications. Not much difference, unfortunately. I even tuned pci latencies to no avail. My card btw is a soundblaster with ensoniq chip, so any obvious driver anomaly presumably would soon be filed as a bug, Fedora Core or otherwise. But, with alsa oss emulation and stuff, it really might be that latencies just add up, which would after all mean that Linux as a desktop system still has it's drawbacks. Anyway, in case you use softphones on Linux, and did compare their performance with Windows alternatives finding that they can compete, may I ask what card/driver/kernel version do work for you? Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability
On Sun, 2005-01-16 at 16:52 -0500, Steve Kann wrote: If the delay goes down after a couple of minutes after the transfer, this could be the problem. Just fyi, this is what I observed with those delays between iaxcomm and firefly, i.e. they occurred on a transfer attempt and normalized after some minutes of talking. Wouldn't be surprised if the transfer was the problem here, too. What I'm not sure about is, due to lack of thorough debugging, whether this is a * or iaxclient library issue ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone for Linux recommendation
On Sat, 2005-01-15 at 05:37 +1100, Howard Lowndes wrote: Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. XLite isn't exactly OSS, isn't it? :) I tried linphone, iaxcomm, gnomemeeting, and SJPhone. Pros and cons: (1) linphone Audio OK, but it doesn't send media when somebody calls me. Buggy for me. (2) iaxcomm Generally good, sometimes crackly audio. When people call me with firefly and * attempts a transfer, we experience huge audio delays in the 10 sec range. Could be firefly or iaxcomm bug. Maybe both, as they both use the iaxclient library. But as long as * sticks to notransfer quite usable. (3) gnomemeeting Fairly good, but currently supports only h323. SIP support is underway. Since I couldn't yet find a simple way to register with * without a gatekeeper and have people call me, I'm currently not using it. (3) SJPhone (for linux) Not OSS, but for me best audio and latencies. Definitive con: there seems to be no dial pad (i.e. dtmf interaction during call not possible). Especially, the OSS phone situation is not really bright, at least for me. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability
On Fri, 2005-01-14 at 16:27 -0200, Denis Galvo - iSolve wrote: Em Sex 14 Jan 2005 16:11, Dan escreveu: I dont have problems when calling PSTN extensions, and calling VoceMail, EchoTest, etc. The problem is related with the conversation between two DIAX Softphones. With * in the middle or direct calls? I had problems with iaxcomm - * - firefly communication when * attempted a transfer. Huge latencies (10 sec or so). Might be a bug in the iaxclient library, just don't know. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] linphone - NAT - * - NAT - firefly woes.
Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid=Foo 100 qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar 10 host=dynamic secret=secret nat=yes canreinvite=no On iax exten 10 I register firefly, on sip exten 100 linphone, both behind nat. Now, calls I can do is e.g. firefly - * - linphone linphone - * echo test (copied this from demo and put it on exten 600) but what wouldn't properly work is is sip to iax bridging linphone - * - firefly More specifically, firefly rings properly, but when I press Accept it just keeps ringing, and finally * tells me that linphone didn't send any frames: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/10-e8bd Jan 12 21:55:02 DEBUG[22661]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/10-e8bd and IAX2/100/2 Doing my tcpdumps I checked that there's really no data sent by linphone, while nothing is dropped by firewalls either. Did anyone experience similar troubles? A hint about how to resolve or further debug this would sure be appreciated. Another point I'm wondering about is why, in that same connection, the caller id handed to firefly is just 10, and not the one specified in sip.conf, i.e. Bar 10. I tested all that stuff also with iaxcomm, i.e. pure iax bridging iaxcomm - NAT - * - NAT - firefly and here, everything works OK, calls in both ways and caller id transmission. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] linphone - NAT - * - NAT - firefly woes.
On Wed, 2005-01-12 at 14:39 -0800, Erik Espinoza wrote: Did you enable passthrough for the rtp ports on the asterisk box? I had the same problem until I enabled udp 1:2 on the firewall. I did. That's why linphone - * echo test works. Maybe I made some progress however, by logging linphone output and comparing the successful echo test to the unsuccessful iax bridge. On echo test I see: (linphone:5450): LinphoneCore-WARNING **: payload PCMA is not usable or enabled. (linphone:5450): LinphoneCore-WARNING **: This remote sip phone did not answer properly to my sdp offer! MediaStreamer-Message: ms_filter_add_link: OssRead,0 - GSMEncoder,0 MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 - RTPSend,0 MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 - GSMDecoder,0 MediaStreamer-Message: ms_filter_add_link: GSMDecoder,0 - OssWrite,0 MediaStreamer-Message: Opening sound card in capture mode with stereo=0,rate=8000,bits=16 MediaStreamer-Message: dsp blocksize is 512. MediaStreamer-Message: Opening sound card in playback mode with stereo=0,rate=8000,bits=16 whereas on linphone - * - firefly: (linphone:5456): LinphoneCore-WARNING **: payload PCMA is not usable or enabled. (linphone:5456): LinphoneCore-WARNING **: This remote sip phone did not answer properly to my sdp offer! MediaStreamer-Message: Mediastreamer processing thread is exiting. I.e. on echo test linphone does select the gsm codec, while with iax bridge the media stream is canceled immediately, hence it stops sending data as properly reported by *. Maybe it's a codec issue, I'm just in the process of investigating ... just thinking, doesn't * transcode between channel legs if necessary, could it be I disabled that by accident (?) ... Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] linphone - NAT - * - NAT - firefly woes.
OK, I'm coming to think linphone is bullshitting me. I now tried the following call paths firefly - * - iaxcomm works firefly - * - linphone works sjphone - * - iaxcomm works, especially sip-iax works sjphone - * - linphone works The opposite paths work too except linphone - * - firefly as said in my orig post, but also linphone - * - sjphone fails. Guess it's about time to contact the linphone people. In case I get that issue resolved I'll post the solution here, too. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
