Re: [Asterisk-Users] Realtime does not work yet, ...
On Mar 11, 2005, at 2:38 PM, Cirelle Internet Products wrote: We bailed on it for now, as it does not appear to be 100%. Phones would not re-register, calls would fail.(just a lot of headaches) I tried out realtime a few weeks ago but stopped using it because it seg faulted whenever I issued a reload command. At least back then it was still too unstable for production. But I do look forward to a more completed realtimehopefully you'll be able to name tables whatever you want then too! Like voicemessages -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Sipura 2100 and Asterisk and Fax
I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty much broken. Finally, this week I decided to spring for a new SIP adapter thinking...err, it must be my old Azacall 200 that's the root of the problem! Boy was I wrong! The SPA-2100 got here a couple of days ago but I didn't get a chance to fully configure it (aka create a dial plan string for my setup) until yesterday. Happily, the first few test faxes to my eFax account went out fine even though I routed them through my Sixtel trunk. For the most part, when I try to fax out through Sixtel I get a microsecond chirp and then silence. At least, yesterday morning everything was working fine and I thought...hey, must be the new adpater! But no. By the afternoon it was back to the chirp and then silence. I switched to testing inbound via Voicepulse. Now that failed too! This had been working time and time again with a hitch with the old adapter? What the?? Must be somewhere in the eight million and growing poorly documented config parameters that Sipura gives you. I already had all of the Fax stuff off and also turned off the echo stuff. No success. I tried changing gain levels...going as low as -12 and as high as +12. No success. I tried all of the 600 and 900 impedance settings. No success. Finally, I thought to myselfmaybe it's timing? Maybe I need to put Asterisk on a faster machine since it's only on an old PPro system circa 1998. Hr. So I installed on an idle PIII that was in the rack. This a.m. I switched over to using the faster Asterisk setup and started running the same fax tests (BTW, I'm using free faxback systems to test inbound faxes, pretty handy. Go google for them!). Started with outbound. Sixtel was still chirping and then dying. So I switched to using VPC's megacent outbound for testing. On the old system I always had to turn trunking off to fax out successfully over VPC so that is the config I went with. Well, it was still no go. I turned all of the FAX stuff back on in the Sipura and then I got a fax to go through successfully. Hrmmm... Next, I tried inbound faxing. No success. Didn't matter if the FAX stuff was on or off in the Sipura. I googled but could find no help on any of this. Most folks seem to still be using POTS for faxing. So I poppped out the Sipura and put my Azacall 200 back in. Sent a test fax. Perfect! Requested several multi-page faxes from a faxback server and they all came in fast and perfect! The Sipura is now sitting on my desk, ready for an RMA#. Unfortunately, the Azacall has a nasty habit of crashing every few days so it's not a permanent solution. What I cannot seem to understand is why the old Azacall works when the Sipura-2100 (brand new hardware, with the latest firmware) just refuses to cooperate in this setup. I figure it must be my fax machine and the baud rate. You see, I have a Brother MFC8300 that's hardwired to 14400bps. I'm thinking that the Azacall will support up to 14400bps (like the later Cisco ATA-186's will) but the Sipura will only support up to 9600bps. I emailed Sipura to find out and all they could manage to tell me was... === You will have better reliability when using 9600bps when using fax. (along with G711, etc..) T.38 fax support for SPA-2100 will be available Q2 === Gee thanks guys! You could just say We're too lazy to ask engineering what the deal is... or We know the real answer but it's a secret... So there you go. It's nice that the Sipura has all of the configurability that it does but, in the end, if there's no way to know how to tweak those parameters or if it still doesn't offer just the right switches and knobs then you're pretty much out of luck. Also, if the hardware itself doesn't support a certain capability (like being able to force a fax machine to retrain to slower speeds) then you're also out of luck. Disclaimer: It is possible that I just have a defective SPA-2100...but what are the odds? Besides, my Azacall has supported T.38 for almost a year now and the SPA-2100 still doesn't have it (at least, not in the current firmware). Not that any of that matters anyhow... -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Realtime does not work yet, ...
On Mar 11, 2005, at 4:00 PM, Matthew Boehm wrote: Here is a perfect example of a lack of user-developer communication. If developers don't hear your problems then we can't fix them. Did you post a bug report? Did you ask for support in -users? IRC? Aparently not cause your problem could have been fixed. Seems like a very Microsoft-like attitude: If it don't work, don't bother to fix it, just go back to what did work. Which res_config_* where you using? What did the debug say? What did the backtrace say? -Matthew Mark Eissler wrote: On Mar 11, 2005, at 2:38 PM, Cirelle Internet Products wrote: We bailed on it for now, as it does not appear to be 100%. Phones would not re-register, calls would fail.(just a lot of headaches) I tried out realtime a few weeks ago but stopped using it because it seg faulted whenever I issued a reload command. At least back then it was still too unstable for production. But I do look forward to a more completed realtimehopefully you'll be able to name tables whatever you want then too! Like voicemessages Well Matthew, no I did not file a bug report even though I do have a login on the bug tracker. In fact, if you hate the fact that I didn't submit a bug you will really hate the fact that I also have a couple of patches for features that I'd like to submit...and I also have a patch for a bug that was in head a few weeks ago but may actually be fixed now. The fact of the matter is it's not that I don't want to submit bugs or features that I like, heck, I'm tired of patching each release that comes out, its just that I've been too busy to review all of the rules and fill out and fax in a disclaimer to boot. I don't think it's the *user's* attitude that's a problem with fixing bugs in this project, giving feedback, etc., part of the blame IMHO is the development process, specifically the policies. They're not exactly motivational if you have several other projects that must take priority. As for support in IRC or in -users? Are you kidding me? The IRC channel is a joke. Any time I've ever gone in there to post a question it gets promptly ignored so people can participate in more off-topic subject matter. The dev channel is pretty much dead. And although I hang around the -users list (this list) you may recall that a certain well-known participant (an asterisk developer, in fact) decided it was more important to insult me about one of my replies to a another user before that well-known poster went ahead and created a web page which pretty much re-iterates exactly what I had written already. So thanks for the inspirational message about why I, and other users, should drop everything else and sign up for even more abuse just because we found a bug. -mark ___ 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] Recommended Phone for beginner
On Mar 7, 2005, at 9:32 PM, Ryan Burke wrote: Hello everyone, I've been watching this list for a while, but it is the first time I've posted. I'ved decided to setup a * server for my house and will need 3 phones (one main, one for my wife, and one for my office). I was wondering if there was a particular brand that people reommended? I'd like ot get an actual SIP phone, instead of an adapter like This question comes up a lot on this list. I mean, A LOT. You'd be better off searching the archives. I'm searching for a low cost IP phone too and have put it off until the Grandstream GXP-2000 is available. Here are the issues as I see it: 1) Sipura 841 ($85): - you either like it or hate it, there are tons of comments about the rubber keys - no backlit screen (in this day age you think everyone would be backlighting by now) - no PoE (if you're going to upgrade to IP phones why bother with power bricks?) - buggy, but it looks like all of the phones are - screenshots I've seen of the screen graphics are not very impressive,,,it looks kind of cheap - screen is not adjustable and the phone sits pretty flat 2) Polycom IP 300 ($139): - no PoE (although it is an option via a special cable that sells for around $40) - no speakerphone (gee, and Polycom is known for their speakerphones...go figure) - you need to upgrade to the IP500 to get a speakerphone - you need to upgrade to the IP600 to get a speakerphone and PoE 3) Grandstream BudgetTone 101, 102 ($75): - these phones are basically cheap and crap 4) Grandstream GXP-2000 ($115 est.): - supports up to 11 lines - built-in 2-port switch - supports PoE - backlit AND adjustable screen - speakerphone - what's not to like? well, it could turn out just to be a fancy BudgetTone at that price point but if it's not then they'll be the ones to beat. - unfortunately, it's unknown when this unit will be available, current estimates are sometime in March. 5) SwissVoice IP10s ($150): - don't hear much about these but it might be just because they only added SIP last Fall (still, they've supported MGCP for quite a while). - those that have 'em seem to like 'em 6) Zultsys ZIP 2 IP Phone ($94): - haven't seen much written about these either - they look pretty cheap 7) Snom 190 ($230): - seems to be highly rated, but it also costs about $230 8) Cisco 7912G ($245): - haven't heard much about these at all - the problem with Cisco is related to getting firmware updates...I think you need an ongoing service contract which makes them more of an ongoing investment rather than a purchase. It would be nice to read more reviews about these products with proper pictures rather than those miniscule ones that all of the vendors seem to provide. Also, you need to keep in mind that just because something is an IP phone that doesn't at all mean it will play nice with Asterisk (or any other specific system). The good thing is that it looks like 2005 will be a great year for cheap, full featured, IP phones. The longer you can wait the more choices you will have. It's always good to start with an analog adapter, IMHO, because if you need fax support you will probably need one of those too. And if you want to add a cheap cordless phone (that is, multiple handsets for less than $400) then you will need an adapter. For now, I've chosen to ride things out with my old Siemens Gigaset and have ordered a Sipura 2100 as the only immediate upgrade (not to impressed with my current adapter). -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] CallerID - Broadvoice vs. VoicePulse
On Mar 8, 2005, at 8:15 AM, Adam Robins wrote: Last week, due to numerous user quality complaints with Broadvoice, I started a VoicePulse Connect account. I've tried both SIP and IAX2 setups, but can't get VoicePulse to pass the callerid from SIP.CONF in similar fashion. I can issue a SetCallerID command before I dial, but it passes number only. Any thoughts? Voicepulse Connect doesn't support CallerID with Name. Supposedly they plan on adding it but if/when they do it will probably be an optional feature that carries an extra fee. Also, any reason to pick IAX2 over SIP setup or vice-versa? I'm still not convinced that inbound DTMF works properly over IAX with VoicePulse. You might want to exercise that capability to see if one gives you better performance over the other and report back here. Supposedly, they know they have a problem with DTMF (and at least when combined with IAX) and that they need to upgrade software. Who knows what the exact problem is and whether or not they upgrade has been made. They don't like to keep customers informed of technical problems and progress in fixing them. Also, I've had not luck faxing out from Voicepulse, at least when I'm using trunk mode. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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: No ringback over IAX - LiveVoip
Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] T.38 fax summary
On Feb 25, 2005, at 11:41 AM, Steve Underwood wrote: Mark, In the time it took to write all that you could probably have read up enough about T.38 to realise you were talking complete rubbish :-) Gee thanks Steve. And your insight has been absolutely beneficial as well. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA that actually work with T.38
On Feb 25, 2005, at 7:55 AM, Steve Underwood wrote: If you understand what T.38 is you will understand which problems it addresses (summary: it is important for solving some problems, but nothing solves them all). Most people who post about T.38 don't actually have much of a clue about it. I think the biggest hurdle still for T.38 is lost packets and timing issues. In other words, the realtime-ness (?) of it is a huge problem. IMHO the whole thing's a bust until we all get QoS across the public network. And let's face it, if you have a private IP network with QoS you really don't need T.38. So I'm a bit lost as to how T.38 is really a solution to much of anything at this point yet the hype would have one conclude otherwise. As for Asterisk not having to know much about T.38...well, that's only true if the only support that will be available (on the Asterisk end) is via an analog adapter that supports T.38. If you want to hookup a fax machine to a port on a channel bank or a zap card then you're going to be out of luck unless the zaptel driver supports T.38. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] T.38 fax summary
On Feb 25, 2005, at 10:20 AM, Lee Howard wrote: In a traditional analog fax you have modulated audio data, that is, the data stream is converted into an audio representation by the transmitter, and the receiver demodulates the audio stream to produce the data stream. A lot of data gets packed into very small portions of audio, which is why fax over VoIP (T.38 is not VoIP, it is FoIP) is unreliable - any jitter will likely cause data loss. There are no modulators in T.38. So take the fax procedure, but instead remove the data modulation/demodulation part. T.38 devices communicate raw data through the IP network, and the IP network is as good at communicating data as the PSTN is as good at communicating audio. So if you could have a full T.38 delivery route from fax sender to fax receiver, the data never once gets converted into an audio signal - it doesn't need to be. Sort of...but no. Fax requires a codec that supports the frequency spectrum of a POTS audio channel. Currently, that means that anything other than g.711 won't work since the other popular codecs achieve their efficiency by dumping frequencies humans can't hear (just like mp3). The problem isn't typically g.711 because that's the codec that is generally used by the digital telco world. A common problem when discussing g.711 often is packet size vs bandwidth limitations. T.38 can alleviate this problem because it doesn't rely on a codec. The bigger problem with faxing over VOIP is related to lost packets and timing issues (jitter). Lost packets are the death knell for fax because it isn't very tolerant of missing data. How do you complete an image with missing data??? AFAIK T.38 can't do anything to recover from packet loss...the fax machine needs to be tolerant of it. Ironically, ECM was introduced to recover from information loss when transmitting faxes over analog lines but ECM can actually cause problems when used with T.38. If you can turn ECM off that's the best thing to do when using T.38. Besides lost packets though if you have to consider packets arriving at weird timing intervals (jitter). The fax machine needs to get its data in a steady stream. This is supposed to be a realtime transmission after all. While T.38 can absorb some of the problems triggered by latency and jitter, when the problem becomes too excessive it tanks just as quickly as faxing without T.38. So with those barriers out of the way what is it that T.38 tries to accomplish? Instead of sending a fax over VOIP as a stream of sampled audio, the protocol intercepts the audio at the endpoints and packetizes it as blocks of data instead. The receiving gateway must know how to handle the data stream so it can convert the fax back into a T.30 fax data stream for POTS. During the session, progress is faked so that the two fax machines don't think the transmission has stopped...that's a crucial step because it takes time to convert and send/receive the fax reliably. I think the best arsenal for faxing over VOIP today is to have a good broadband connection, g.711, and a fax machine where YOU can set the max transmission speed. Sadly, the last part seems to be missing quite often. I've noticed that HP actually mentions faxing over VOIP in the documentation for their 7410 all in one machine and, more importantly, they include support for changing transmission speeds. Way to go HP! -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA that actually work with T.38
On Feb 24, 2005, at 3:05 PM, Brian M. Arlinghaus wrote: So... If Asterisk did support T.38, would that solve the world's fax problems? No, because T.38 must be handled all the way through, up until the final gateway that will take the T.38 session and convert it back into POTS. The world doesn't really have a fax problem per se. Fax was developed for analog lines. What the world needs is fax machines that are completely IP-based and only fall back to using a gateway when connecting to POTS-based fax machines. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] FAX
On Feb 20, 2005, at 1:26 PM, Brian Roy wrote: On Sun, 20 Feb 2005 23:16:00 +0900 (JST), Isamar Maia [EMAIL PROTECTED] wrote: Ok. I will be burned in fire.. :-) Or better.. I won't go to the heaven... You are probably right. But in the the mean time, while you are here on earth, you will probably spend some time in the legal system too. Spam faxing is a punishable offense and enforced per incident. War dialing for fax machines fall under the same category. Spend a little time here before you get too far into the project. http://www.junkfax.org/index.html If you impede someone's ability to get the e911 system by clogging their lines that goes beyond illegal. Find another get rich quick scheme. -Chuji E, let's keep in mind that the original poster may not be located in North America. Either way, I'm sure the OP got the message that they won't find help here. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Minimal hardware requirements
On Feb 21, 2005, at 7:35 PM, Rudolf Ladyzhenskii wrote: Hi, all I am doing prrof of concept system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate? Scrap any analog connections. Get a VOIP SIP adapter to handle analog. Setup VOIP to PSTN termination via one of the many providers. No need for TDM cards. Your system will work fine. I've done this on an old Pentium Pro with only 128MBs for a small system. You only need bigger hardware if you're going to add many more users. Two SIP phones is nothing. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] fax with asterisk
On Feb 17, 2005, at 2:32 PM, Justin Richards wrote: I don't do a lot of faxing, but I would like to know I'm going to receive them when I do get one.. I think therein lies the key to your problem. If you're not doing a lot of faxing then its hard to know if the problem is at your end or if its somewhere else (like your ISP). Sending or receiving a fax every now and then means you can easily fall into this situation: 1) after a successful fax or two during a given period of time you think AHA! this is working without a hitch. 2) after an unsuccessful fax or two during a given period of time you think DANG! this is no longer workingwhat changed at *my* end? At least that has been my experience using several VOIP providers but always the same ISP with no changes at my end. It comes down to the fact that some people are lucky to have an ISP that maintains their network properly while others must suffer at the hands of an ISP that does a really crummy job. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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, inband DTMF send by a GSM mobile
On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote: I find the compilation option RADIO_RELAX. this option change a threshold in DTMF detection (function dtmf_detect in dsp.c) I remark an big improvement in the detection of the dtmf over GSM. have you ever test this option? RADIO is obscur for me, does it mean all wireless device? I haven't dug into the source looking to fix this problem so, no, I haven't tried enabling that option. Perhaps someone on the list knows more about this option? I'll certainly try it out as well. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA that actually work with T.38
On Feb 15, 2005, at 2:32 PM, Matthew Boehm wrote: Stop. The PAP2-NA's have no T38 support. Next time, lets try and read the OP's message before responding. -Matthew Hah! With over 2-- or 300 messages per day we're supposed to read everything in them? I find it easier just to respond to multiple posts in long response anyhow! ;-) -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA's
On Feb 15, 2005, at 6:08 PM, Leo Ann Boon wrote: From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. Guess I never really looked at it that way. Perhaps when if I cancel my Vonage account I'll just hang on to the ATA [he casually comments as the collection of VOIP adapters steadily grows in the basement...]. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] DTMF inband detection improvement
On Feb 16, 2005, at 10:01 AM, Peter Svensson wrote: On Wed, 16 Feb 2005, Steve Underwood wrote: If you really are using ulaw, and you do not have extreme packet loss or jitter, DTMF detection should be very reliable. It is no better in CVS HEAD because it wasn't broken in the first place. We have some problems with dtmf detection on our lines. We use E1 pris connected to a TE405P. Mostly we see duplicated digits. Unless the signal is perfect and distortion free Asterisk sees the small imperfections as the end of the digit. We can provoke this problem from some of our office phones (when on speaker phone). Asterisk is more or less the only place we see this problem. I concur. I have DTMF problems with inbound calls over IAX. Don't have any DTMF problems locally using g.711. I also have problems with inbound calls from GSM phones but hey, that's not a surprise and yes, if you dial realy slow then it seems to work more reliably. BTW, Steve, if you're still reading, what is the RADIO_RELAX option intended to be for in dsp.c? -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] DTMF inband detection improvement
On Feb 16, 2005, at 10:34 AM, Steve Underwood wrote: BTW, Steve, if you're still reading, what is the RADIO_RELAX option intended to be for in dsp.c? It is something someone else added to the code to make the detection criteria in relaxed mode even more relaxed. If setting that helps, something in your channel must be causing some serious filtering of low frequencies. Can you try logging the audio to a file, and send it to me for analysis? chan_spy, or something like that, should do the job. Actually, it was Florian that posted about this option. I haven't tried it (spent an awful lot of time last week compiling different configurations of stable, head, patches...taking a break this week). This is Florian said: On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote: I find the compilation option RADIO_RELAX. this option change a threshold in DTMF detection (function dtmf_detect in dsp.c) I remark an big improvement in the detection of the dtmf over GSM. have you ever test this option? RADIO is obscur for me, does it mean all wireless device? Florian -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Sipura g729 call quality to PSTN
On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA that actually work with T.38
On Feb 14, 2005, at 5:27 PM, Cory Andrews wrote: There is just a form that needs to be completed, which we forward on to Linksys and they approve or deny the application based upon the background of the applicant. Have had very few applications rejected, pretty straightforward process. I don't get it. My assumption is that by background you mean the applicant turns out to be a VOIP provider as opposed to say a lot of people on this list that would be interested in buying one of these devices but can't because they're really just end users. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA's
On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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, inband DTMF send by a GSM mobile
On Feb 15, 2005, at 10:09 AM, Florian Lefeuvre wrote: Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this I suffer from this problem all of the time but I'm configured for out-of-band dtmf on the asterisk side thanks to IAX2 trunking. I think the problem is GSM or maybe cell phones in general. I know this has been discussed before. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Sixtel.net / IAX.CC - Vanity Toll-Free Number
On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote: I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. Well let me tell you one thing about that, whenever a VOIP provider runs out of DIDs in their pool, it can take days. I had the same problem with Voicepulse several weeks ago where I ordered two DIDs in the same exchange. Got the first number immediately but the second one took somewhere between 4 to 6 weeks. Come to think of it, the exact same problem happened to me with Broadvox almost exactly one year earlier. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] soho fax suggestions?
On Feb 13, 2005, at 4:43 PM, John Novack wrote: I use JFAX which I think is also known as Efax. If you are open to a new fax number anywhere else in the US from your home Zip code, then it is free. Otherwise there is a quarterly fee. AFAIK, you can't port an existing number to them, but I could be off on that. http://www.j2.com/jconnect/twa/page/servicesOverview I have a free eFax number that I've maintained for testing...although I'm unable to fax to it via Sixtel (you begin to hear a carrier but within 1/2 a second it's cut off). So much for testing. I have also used a Broadvox residential account for inbound faxing (they include fax-to-email as part of their feature set). But I think they may have broken this feature recently when they switched to a new VM system. While you might not be able to port a phone number to eFax, there's nothing stopping you from forwarding a number to eFax. But like I said, I've found outbound fax to be more of a problem than inbound. While the latter has worked well for me with Vonage and Voicepulse, the bigger problem is the former (outbound) as it's only ever worked reliably for me with a plain residential single-line account that I've had since May 2003. With Broadvox faxing was completely unreliable and often didn't work EVEN THOUGH they have T.38 support. Here's what I learned though: just because your CPE supports T.38 and your provider's gateway supports T.38, that doesn't mean that the carrier sitting in between supports T.38. Level 3, for instance, doesn't support T.38 at the moment (at least, not in all markets). So IMHO, T.38 ain't gonna do anyone any good until it's implemented across the board and who the heck knows when that might happen. While eFax, and similar services, are some sort of a solution to at least half the problem, I just think using these services is a kludge. The beauty of fax is: stick a document in at one end, dial a number, and the document spits out at the other end. No clumsy scanning and emailing involved. And while some folks think Fax is dying, I just don't agree. I think the technology needs to be rebuilt for IP, but I don't think the concept is going to go away anytime soon. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] soho fax suggestions?
On Feb 13, 2005, at 7:50 PM, Rich Adamson wrote: Can't offer any clue on the above either. Based on Steve Underwood's comments earlier (relative to outbound fax now fails on the TDM when it was working earlier), it would almost sound like a timing issue of some sort that is associated with calls initiated within *. Interesting. I wasn't aware of that. I'm more inclined to blame my CPE at the moment. Will probably switch to a Sipura 2100 soon. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] ATA's
On Feb 14, 2005, at 5:39 AM, Nicolas Bougues wrote: On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote: We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2 for the SPA-2000. I think Matthew was referring to the lack of leds on the front of the Sipura. I can't seem to figure out why these manufacturers insist on building these boxes like you're going to stick them on your desk next to your phone. I want something that's more suitable for a phone closet. Too bad the PAP2-NA can't be purchased retail anymore. Then again, you're probably better off with a Sipura-branded unit anyhow. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] soho fax suggestions?
On Feb 12, 2005, at 4:38 PM, Rich Adamson wrote: For planning purposes, is it appropriate to think in terms of purchasing a t38 capable box even if its not supported by * today? (I'm well aware of the bounty and Steve's work.) That's what I would do. In fact, I already have T.38 capable VOIP adapter (an Azatel 200) for my current fax machine but plan to upgrade that box to a Sipura 2100. If now is the time to purchase a t38 capable fax machine, anyone have any suggestions on a low-volume soho-sized box? I don't think there is such a thing as a T38 capable fax machine. T38 is for faxing over VOIP and I have yet to see a fax machine with a built-in network port so it can connect directly to the Internet...if you know what I mean. FWIW, I have had absolutely zero problems receiving faxes over VOIP, via Asterisk, using Voicepulse Connect, IAX trunking, and g.711. My problems for faxing are all related to outbound faxing (using the same Voicepulse setup or Sixtel iax.cc). Not sure why outbound is giving me problems when inbound isn't giving me any. shrug Of course I need to fax outbound more often than I need to receive inbound! -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Is there a Caller ID issue in the latest CVSStable
Wait a sec, so let me get this straight... Previously if you checked out cvs 1.0.5 stable you would get the caller id bug and now there's a 1.0.5 stable that doesn't have the bug? Errr, someone please explain to me the versioning scheme being used here. Seems to me that if stable is released with a bug then the only way to change that is to issue a new release (like 1.0.6) without the bug. -mark On Feb 10, 2005, at 5:49 PM, Nicolás Gudiño wrote: Paul, 1.0.5 stable suffers from caller id issues as well, at least for SIP channels. What fixed things for me was swapping in app_dial.c from 1.0.2 stable (didn't try others). You could also just diff app_dial.c between versions to find the problem but I took the lazy way out the first time around. Drumkilla reverted the callerid changes on the latest stable (thanks Russell!). You will be fine if you checkout stable from CVS now. Regards, -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] dtmfmode and IAX protocol
There's enough information if he's using FWD's 8XX-gateway for his toll-free calls to UPS, the bank, etc. First of all, if the wiki says inline (yes, okay, it does) it probably means as inline data as opposed to inband. But the fact is that iax2 ALWAYS sends DTMF out-of-band. While it's true that some VOIP phones (SIP adapters, etc) can be configured to send DTMF inband, I would think that doing so while using IAX is going to result in digits being reproduced twice at the destination--once from the DTMF reproduced from out-of-band and once from the inband DTMF. So if you're using IAX as your trunking protocal you need to use out-of-band DTMF on your IP phones (and adapters) as well. Why does DTMF work sometimes and not all of the time? Heck, if I knew that then I wouldn't have this problem where inband DTMF hardly ever works properly for any of my inbound calls over IAX from Voicepulse. I'm starting to think that Asterisk's support for DTMF over IAX has issues but I'm too stubborn to switch to SIP and test that. I know I don't have any (zero, nada, keine, rien, etc.) problems navigating Asterisk IVR menus via my SIP adapter. It's important to keep in mind, however, that the telco environment beyond your Asterisk box, beyond FWD (Voicepulse, Broadvoice, Vonage, etc.) is a complicated environment where everyone isn't playing by the same set of technical specs. The fact that any of this stuff actually works as well as it does is just amazing in itself. -mark On Feb 11, 2005, at 1:17 AM, Rich Adamson wrote: Joseph has been working at bringing up an asterisk box as kind of a newbie, and I think he's using a Sipura as his fxs interface into asterisk. He's having a problem with asterisk passing dtmf to FWD, but didn't say how he's accessing the bank or fedex. So, without a fair amount more detail from him, there's no way to answer his questions or guess at the problem. Exactly. (I was hoping he'd come to his own conclusions.) So... if the Sipura does not do IAX, then it's quite possible that you're not doing IAX on the Sipura. Which means the whole dtmfmode and IAX protocol is moot... -Michael - No. Can the Sipura SPA-3000 do IAX? -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Thursday, February 10, 2005 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol Actually, I don't know what might be the problem. I'm using Sipura SPA-3000 unit connected to standard cordless phone and connecting to FWD over IAX 1.) If I call FedEx or Bank and enter my account number using numeric keys it works 2.) If I dial UPS 1-800-742-5877 and try to use one of the option provided it doesn't work. Could it be their phone system? -- #Joseph On Thu, 2005-02-10 at 21:36 -0600, Michael Giagnocavo wrote: Actually, there are some phones that will do inband DTMF over IAX2. So if he's using one of these, he has to make sure his settings are correct. Yes, the PA168 phones. The correct setting is RFC2833 for IAX (inside these phones). Otherwise it's inband. The other options they provide just cut the call. -Michael ___ 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 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 ---End of Original Message- ___ 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 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 ---End of Original Message- ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
I'm confused about the part that you can check out a stable version after 1.0.5. IMHO if you check out what is tagged as 1.0.5 at any time then you should get exactly what is in the 1.0.5 tarball. If you check out head then you should get all of the latest stuff in CVS which may or may not build cleanly (and may segfault or whatever). If you could check out 1.0.6-rc1 (release candidate 1) or something like that you would get everything after 1.0.5 that may or may not build properly but is no longer a moving target (features have been frozen). It just doesn't make sense to me that there would be a 1.0.5 that has changed since 1.0.5 was released unless you tag it 1.0.5.1 (or something). I mean, why even bother trying to constantly maintain a new stable version without having a formal release? 1.0.5 is what it is with whatever bugs it came with upon release. Obviously, just my opinion on How things should work!. ;-) -mark On Feb 11, 2005, at 10:06 AM, Chris Wade wrote: It basically works out that CVS *is always a moving target*, tarballs are the only things that don't change. Make sense? -Chris -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 Voice Quality Issues
There's a new jitterbuffer in development right now for IAX. To use it, however, you need to patch cvs head. -mark On Feb 9, 2005, at 1:49 PM, Paul Rodan wrote: That jitter buffer has caused nothing but problems for me. But that was a few months ago, haven't tried it lately. What are you using as your timing source? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Dingman Sent: Wednesday, February 09, 2005 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX Voice Quality Issues Silence Suppression is off or set to no of the SPA. I changed jitterbuffer=no and things seem better. Will need to do some more testing. On Wed, 09 Feb 2005 13:03:06 -0500, Andres [EMAIL PROTECTED] wrote: Just as further info, I am using a SPA-2000 to connect to * with G711u as the preferred codec. Maybe you have silence suppression enabled on the SPA? That does not play nice with Asterisk. ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Is there a Caller ID issue in the latest CVSStable
Paul, 1.0.5 stable suffers from caller id issues as well, at least for SIP channels. What fixed things for me was swapping in app_dial.c from 1.0.2 stable (didn't try others). You could also just diff app_dial.c between versions to find the problem but I took the lazy way out the first time around. Caller id seems to be fixed in the current cvs head but for production you should probably stick with stable. -mark On Feb 9, 2005, at 12:31 PM, Paul Rodan wrote: I thought that as long as I stuck to the stable branch, only major bug fixes would be included, no new features or changing of the way things are handled? I mean, isn't the latest CVS Stable better than 1.03? I'm in the asterisk-cvs list and every day I see bug fixes added to the stable branch that fixes segfaults and divide by 0's and typo's here and a mistake there, etc. etc. won't all those bugs be present in the 1.03 version? I don't want my system to seg fault as the cvs list would indicate it could. So there are known issues with the latest CVS Stable? What is the best known version of Asterisk to date? 1.03? 1.05? I'm not interested in new features, just stability and quality. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] SIP jitter?
I've discovered that one of the pitfalls of wanting to try out the new jitter buffer is that you have to move to CVS head... Which isn't a biggie unless you've been using mysql without odbc. Am I dreaming or is the old type of non-odbc sql support eliminated from cvs head? Anyhow, just thought I'd put that out there as a warning for anyone considering patching head for the new jitter buffer. -mark On Feb 8, 2005, at 8:25 PM, Andrew Kohlsmith wrote: On February 8, 2005 07:53 pm, Steve Kann wrote: Glad it's working for you, Peter.. Seems to be working for me too; I'm using both 2532 and 3400. Your iax2 test pktloss patch moved my build to /opt/asterisk/vCVS which caused me some consternation but it's all good now. :-) -A. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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: iax2-jitter-trunking?
The wiki also shows tests with trunking on/off and using different codecs including g.711. It's not stated anywhere that g.711 doesn't support it: http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 With that said, I'm pretty sure I read it too somewhere that the codecs have to support trunking and that g.711 doesn't. Perhaps it was on the list somewhere? -mark On Feb 7, 2005, at 11:44 AM, Kevin P. Fleming wrote: joachim wrote: The codecs dont need to support trunking... Ahh, I could have sworn I read that on the wiki, but now I can't find it. Must have been Monday morning brain failure :-) ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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-jitter-trunking?
AFAIK, trunk=yes is not a global option. You set it within a context.
Also, using the jitter buffer with trunk=yes is not recommended since
its broken right now.
-mark
On Feb 6, 2005, at 12:45 PM, Rich Adamson wrote:
Two cvs-head asterisk boxes with iax2 working fine (without register
statements).
When two calls are placed simultanously from system A - B and the
packets
are sniffed on the wire, I see the two calls using two different udp
packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes
(at both ends).
I was expecting to see both calls handled within a single udp packet,
but that's not happening. Each iax2 packet is 79 bytes using ethereal.
I've tried the trunk=yes both within the inbound context and at the top
of the iax.conf file (assuming the one at the top would be used for all
outbound iax calls that don't reference a context). Calls are placed
with:
exten = _2.,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
Is trunking dependent upon the use of 'register'? Or, dependent on the
above exten=_2., referencing a context (instead of the IP directly)?
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
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Re: [Asterisk-Users] Callerid problems with 1.0.5
I saw the note about the o flag. But if I'm not mistaken we have to add the o flag to get the old behavior back again instead of using the o flag to get the new behavior. That means our existing (and formerly functioning) dialplans will be broken. It would make more sense IMHO to have a flag that turns on the new behavior rather than having one that restores the old behavior. But then again, I also think that if a call is bouncing out of call park that the caller id should indicate that as opposed to presenting it as a brand new call. -mark On Feb 4, 2005, at 9:35 PM, mattf wrote: Hello, patching v1.0.5 on my system removed the problem for me. But yes it seems strange that this feature was inserted into a final release with very little documentation of the wide implications that are caused by the change. This was corrected in CVS with the addition of a diabling flag for the dial command, but maybe this is a message that we should start an official beta release period before a release so that people can test pre-releases even for just a week to report problems before it is unleashed upon the world as an official release -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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: No ringback over IAX - LiveVoip
I'm quite happy with iax.cc (Sixtel). I don't have DIDs with them but use them for outbound and have no complaints. Whenever I've contacted support I've received a reply the same day. Perhaps they prioritize their support email based on whether or not you have an account with them? They match up inbound support messages with your email address to check if its associated with an account. Anyhow, I might obtain DIDs from them sometime soon if Voicepulse doesn't get their act together and gets inbound DTMF working properly over IAX. It's been almost two months since I first reported the problem and they just reply that they have to perform several software upgrades over a 2-4 week period. Jeez. I can't recommend Voicepulse either right now as I have no intention of switching back to SIP (from IAX) for termination. The funny thing is that I have ended up using FWD the most because of their toll free gateway. I'm constantly amazed at the clarity of those calls. But then again I'm constantly amazed at the clarity of any of my calls through Asterisk vs. say my residential phone services via Vonage and Broadvox. -mark On Feb 3, 2005, at 10:38 PM, Brian Dingman wrote: I took them up on their offer for a refund. IMHO they shouldn't offer * service at all. Even outgoing calls aren't handled properly. Lots of making progress - no answer results. Others have suggested iax.cc. However, they haven't repsonded to my email (over 2 days now) and I can't get through to them over the phone or IM. Not very promising. All I want is a toll free DID that works on * and isn't too expensive. Any suggestions for a provider? I don't even care if it can be ported away! On Thu, 3 Feb 2005 10:12:02 -0500, Mark Eissler [EMAIL PROTECTED] wrote: Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Callerid problems with 1.0.5
Yikes! On Feb 4, 2005, at 1:26 PM, Jay Milk wrote: Can someone clarify what's going on here? I'm running 1.0.5, and I see caller-id come through just fine from one extension to the other, as well as for incoming and outgoing calls (iax2). What are you folks seeing there? The behavior that was reported by Kevin is/was exactly the same behavior that I was experiencing with 1.0.5 and reported in another thread. I switched back to 1.0.2 to resolve that problem and another I was experiencing (SIP calls ringing forever instead of disconnecting even when voicemail had already picked up). Reading through the bug tracker on this one I must say I'm a bit confused. I understand the concept of showing useful/relevant callerid when a call is transferred (from park or some other extension) but I don't understand why a call should ever show the recipient extension's callerid. My understanding is that this is the default behavior when no other callerid is present and for some reason inbound callerid is getting wiped out because it's not correct. That some people are experiencing problems with this while others are not leads me to believe that it might be a problem that is exacerbated depending upon the dialplan setup. I'm just thinking this at the top of my head now, haven't looked back at my dialplan yet. What's annoying, either way, is that when this change was made the behavior of existing, functioning setups broke. I don't recall seeing any documentation for 1.0.5 that noted this might be the case and if the documentation is lacking...well, that's a problem. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Mi extensions keeps ringing
I seem to be experiencing the same problem (I'm using IAX trunks). My SIP phone keeps ringing after a call has been disconnected. This is with * 1.0.5. I'm preparing to back out to 1.0.2 where I wasn't having this problem. Also, the callerid info passed to the called extension is of the called extension itself instead of the caller. Weird. -mark On Feb 3, 2005, at 4:00 PM, [EMAIL PROTECTED] wrote: Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, Ive configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows: -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Mi extensions keeps ringing
Okay, going back to 1.0.2 fixed this problem. Looking at verbose feedback I see I was getting 481 errors as well (Call Leg/Transaction Does Not Exist) using 1.0.5. I have an Azatel AzaCall 200 SIP adapter. So something changed between 1.0.2 and 1.0.5 to make this happen. I didn't see a bug report for this in Mantis. Is anyone else experiencing these problems?: 1) incorrect callerid; 2) SIP extensions not disconnecting? -mark On Feb 3, 2005, at 6:05 PM, Mark Eissler wrote: I seem to be experiencing the same problem (I'm using IAX trunks). My SIP phone keeps ringing after a call has been disconnected. This is with * 1.0.5. I'm preparing to back out to 1.0.2 where I wasn't having this problem. Also, the callerid info passed to the called extension is of the called extension itself instead of the caller. Weird. -mark On Feb 3, 2005, at 4:00 PM, [EMAIL PROTECTED] wrote: Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, Ive configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows: -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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: No ringback over IAX - LiveVoip
Based on the support and management responses that have been posted to this list it doesn't sound to me (at least) like LiveVoip really wants business from * users anyhow. They blame a lot of problems on * and are quick to offer a refund. There are plenty of DID providers that are more asterisk-friendly. -mark On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Trunked IAX or not
You must set trunk=yes in the context of the relevant provider. Not all providers support it. The benefit of trunking grows exponentially with the number of calls in progress. -mark On Jan 31, 2005, at 2:24 AM, Spencer Nassar wrote: The test results that Philipp pointed out show some protocol comparisons that include iax2 trunking / alaw and iax2 / alaw and concludes that IAX2 trunking is more than twice as fast as non trunking IAX. Forgive the newbie question, but what is this distinction? In what cases is a connection 'trunking' or 'not'? If I have a register = statement in my iax.conf file, is that a trunked connection to my DiD provider? Thanks! -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Trunked IAX or not
Yes, you are correct. Instead of separating the RTP streams for each call, trunking combines them in order to save on overhead. -mark On Jan 31, 2005, at 9:38 AM, Andrew Kohlsmith wrote: On January 31, 2005 08:57 am, Mark Eissler wrote: You must set trunk=yes in the context of the relevant provider. Not all providers support it. The benefit of trunking grows exponentially with the number of calls in progress. Isn't it just a linear savings? 1 call: UDP overhead + voice data 2 calls: UDP overhead + voice data + voice data 3 calls: UDP overhead + 3xvoice data etc... without trunking the UDP overhead is repeated for each voice call -A. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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.cc / sixtel are they legitimate?
Been using them for just over a month for all outbound calls. Their customer service is prompt and courteous. I use Voicepulse for inbound and turn around for the average ticket I open is about three days whereas for Sixtel it's often been same day. As for the 800 numbers...I don't have one. But if they're charging $0.30 now for an auto-assigned number then their rates have gone up because I'm pretty sure those used to be free. How do they make their money on those? Well, I would think that since most 800 numbers are now recycled it's fairly likely that you will get many wrong-number calls and if you have an IVR running those calls will start to add up at $0.02/min!! -mark On Jan 27, 2005, at 9:21 AM, Jon Gabrielson wrote: Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated. Thanks, Jon. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Anyone having problems with LiveVoIP?
Regarding those comments below. I am not surprised at the answer and I doubt anyone would be that's taken a look at the * code...it's just not the most elegant thing in the world is it? No point in huffing and puffing about inline comments though when you'd have to find and convince who knows how many contributors to the source to use sound documenting techniques. Sigh. FWIW Voicepulse has ongoing problems with Asterisk as well. Now, I would think that VP would contribute whatever patches they find are necessary for the correct operation of *. And I would only hope that LiveVOIP would do the same otherwise they'll have to go on fixing the bugs again and again and again. As an aside: It's pretty bad corporate policy (from a PR perspective) to take a confrontational stance to one's current and potential customers in a public forum. It sure ain't gonna sell anyone on your service. -mark On Jan 27, 2005, at 12:18 PM, Brian Dingman wrote: From Support: Asterisk is full of bugs and in many cases you fix one thing only to have another show up. We suggested users move to 1.0.3 Our team will look at more things in the software as a part of our ongoing support to clients. We are looking at this version as well as 1.0.3 for some other issues now but, Asterisk is not our only platform. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] DTMF digit dropping
Just to clarify Paul, you're connecting to BroadVoice with SIP right? Does BroadVoice now support out-of-band DTMF? DTMF works for me occasionally (over IAX) on inbound calls over VoicePulse. Whenever I achieve success I get all excited and think maybe they fixed it. But then a few more tests and forget about it. Outbound DTMF always seems to work. Maybe it's time to look at the DTMF code in Asterisk. -mark On Jan 26, 2005, at 10:23 AM, Paul Rodan wrote: I have a small IVR on my Asterisk server connected to BroadVoice, I always used DTMF, but I tried to switch to rfc2833 the other day out of curiosity and interesting enough, when I called into my IVR w/ my cell phone, it recognized 1234 and whatever other digits I entered. So inbound DTMF worked using ULaw, however I never tried outbound. Could have been a fluke though. Give it a shot. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 make channel busy signal?
Call waiting? -mark On Jan 26, 2005, at 6:09 PM, Joseph wrote: When I make a call over the Internet and call myself IN over POTS my phone rings to outside party but I can not hear it. Why isn't my channel extension indicating busy status when I'm making call over Internet? This way I could ring my next extension with n+101 priority. I'm using Sipura-3K unit. -- #Joseph ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Directory service of voicemail extensions
AFAIK it does not currently support that. IIRC it actually states somewhere, perhaps on voip-info, that once you enable voicemail db support you will break the directory listing feature. -mark On Jan 27, 2005, at 9:34 AM, Jagan Mohan wrote: Hi, Does Asterisk support Directory service of voicemail extensions using database? If yes, how to configure asterisk? I know that it supports this feature using conf files. Thanks, Jagan ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Anyone having problems with LiveVoIP?
If the problem is with asterisk userswhy is LiveVoip trying to change something at their end? -mark On Jan 26, 2005, at 10:33 AM, Tim Lewis wrote: LiveVoIP did not issue any end user patches last night. They had a problem connecting to Level 3's network. LiveVoIP claimed the problem was with asterisk users, I have not upgrade or install any patches and all is fine now. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Codec negotiation
Can't you just create a different context for inbound and outbound calls? Then specify your codec preference order in there. I don't think you can specify the bandwidth= parameter within a context other than the global one though. -mark On Jan 25, 2005, at 6:13 PM, [EMAIL PROTECTED] wrote: I don't want that... because - for outbound calls I want priority to be g729 first - for inbound calls I want no priority at all (e.g. the calling asterisk to decide which codec we will use) The last doesn't happen.. This by the way DID happen correctly with previous versions of asterisk (1.0.3 for example) the current CVS-HEAD version doesn't -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohammed Salim Sent: dinsdag 25 januari 2005 22:10 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Codec negotiation The order matters in asterisk so if you want GSM to take priority over G729, simply put that ahead of the G729... so your settings should be: Allow=all Allow=gsm Allow=g729 Allow=ulaw Allow=alaw Try that and see if it works. Regards, Mohammed Salim EZZI Telecom, Inc. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] DTMF digit dropping
I'm having the same problem with Voicepulse connect using IAX2. So, no, it's not better IMHO. And I've been thinking about switching to SIP to see if the problem goes away (I'm very reluctant to do so though) but it's hard to know if the problem lies with Voicepulse (or Broadvoice in your case) or whatever CLEC terminates your inbound number. FWIW I have experienced the problem with Asterisk 1.0.2 and now also with 1.0.5. It doesn't seem to be an Asterisk problem though because the vast majority must not be having any issues with DTMF recognition. -mark On Jan 25, 2005, at 8:55 PM, Bryce Nesbitt (mailing list account) wrote: I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions. Is http://connect.voicepulse.com/ , using IAX, any better? ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Anyone having problems with LiveVoIP?
It would be really nice to see whatever patches they develop for Asterisk or at least get some hint of where the problem lies. -mark On Jan 25, 2005, at 9:41 PM, Brian Dingman wrote: LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If you are not using that version you need to upgrade now. We have a problem with two of our carriers at their gateway related to the Asterisk users. Our staff has developed a patch that is being tested at this time. Once the patch has been approved on our testbed we will move it on to the production switch environment. We do not do upgrades like this during the hours or 9 a.m. - 7 p.m. EST due to high traffic loads. We expect to do switch updates after 7 p.m. this evening that should resolve the problems you are having. LiveVoip engineers are also looking at a DTMF problem in the Asterisk software ver. 1.0.3 which may or may not involve you. Both of these issues are Asterisk software related in nature and not LiveVoip LLC switching defects. Thank You in Advance for your understanding. This issue has been placed under a master ticket for tracking. ** When contacting LiveVoip LLC Support please provide us with the latest version of Asterisk you are using, any and all logs if necessary and as much detail regarding any problems you are having. Network Operations Team LiveVoip LLC On Tue, 25 Jan 2005 20:11:43 -0600, Tim Lewis [EMAIL PROTECTED] wrote: Thanks Jeff! I think it's a little too late to find this info out. 3 to 4 days of no service. I have send many emails and still awaiting a response. Reminds me of my ILEC (QWEST) Do you have any info on what this patch does? -later On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote: They are coming out with a patch for the DID problem tonight. Need to have Asterisk 1.0.3 ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] LiveVoip DTMF Issues
My assumption is that most folks trunking through Voicepulse Connect must be using SIP since I haven't seen this problem mentioned before. So my conclusion is that DTMF and SIP and VPC work fine together BUT then you don't get to benefit from the efficiency of IAX. So the million dollar question is: Does IAX have a problem with DTMF or is it just certain carriers that have problems with DTMF? -mark On Jan 24, 2005, at 6:50 PM, Juan Cardenas wrote: I have experience that problem on numerous ocassions with Voicepulse Connect service using IAX for inbound service. DMTF times out or fails to read certain digits(tones). When had it configured to use SIP for incoming calls, it never failed. - Original Message - From: Mark Eissler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Brian Dingman [EMAIL PROTECTED] Sent: Monday, January 24, 2005 3:06 PM Subject: Re: [Asterisk-Users] LiveVoip DTMF Issues Same problem I'm having with VP Connect. Perhaps it's a question of the version of Asterisk being run. I'm on 1.0.2. -mark On Jan 24, 2005, at 11:36 AM, Brian Dingman wrote: I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at all. Is there anything I can do on my end to fix this problem, or is the old axim you get what you pay for true? It should also be noted that I have some other DID's from other providers and DTMF recognition is pretty much dead on. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Some more hardware and E1 questions
IMHO hardware RAID trumps software RAID. In order to use the latter your system must still be operational to some extent. -mark On Jan 24, 2005, at 11:10 PM, Gary wrote: better solution rather than have a machine with raid is to investigate ISCSI :-) On Mon, 24 Jan 2005 09:40:10 +0100, Daniel Nystrm wrote: I will be using Debian, and as long as the Linux Kernel supports the SATA controller, the rest shouldn't be any problems. If it's SATA RAID, I probably will use ordinary Linux software RAID, since it's more powerful than the simple one in the controller. - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 6:13 AM Subject: Re: [Asterisk-Users] Some more hardware and E1 questions Daniel Nystrm wrote: Hi again folks! ;) As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600. Now I'm into choosing server platform. And the two opponents are: * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1) * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1) If you're planning to use SATA RAID on PE750, make sure your Linux distro supports. Your best bet - use Redhat Enterprise Linux or one of it derivatives. I'm using Centos 3, it autodetects the RAID whilst Mandrake 10 failed. As I've seen people having problem with HP server, I havn't looked at it at all. What experience do you have with the alternatives above? Which would you recommend? And another question at the same time; what's really E1? How is E1 devices connected? Seems like regular Cat5 cables, but it problably ian't? If anyone's using Adit 600, did they send all cables required for connecting to the FXS channels? Seems like a very unique plug on the side of Adit. Thanks! BR Daniel Nystrm ___ 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 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] LiveVoip DTMF Issues
What does it say about * and providers? Err, I dunno but the whole issue is giving me a splitting headache! Is everyone else using g.711 too? This is my setup: -Asterisk 1.0.2 -IAX (currently set to trunk=no) to VPC -codec is g.711 -tos bits are 0x18 (low delay, high throughput) -jitterbuffer=no Is my understanding correct that with IAX dtmf is always sent out of band regardless of the codec selected? Question: Why do you suppose only one line is okay with LiveVoip (with regard to DTMF)? It must be something outside of Asterisk that's causing the problem. Voicepulse doesn't really get too specific when they acknowledge a problem though. -mark On Jan 25, 2005, at 9:15 AM, Brian Dingman wrote: Mark, I don't know what to tell you. With my DID's from VP Connect, DTMF works fine over IAX. Even one of the lines I have with LiveVoip seems OK over IAX. The other well... it really doesn't work at all. So what does this say about * and DTMF recognition over IAX? Or the service providers? -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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: Bellster - IAX-based interchange -- lets you call anywhere for free
It's completely legitimate for the telcos to monitor this type of thing. I would think that the primary users of the Bellster system would be home users and residential lines are not meant to be tied up full time. That's how capacity is determined and that's how rates are calculated (based on usage). It's the same problem that Mom Pop ISP's ran into while trying to run out of a basement. Besides, telcos are going to be wary anyway of this type of thing because of the erosion of their traditional business thanks to VOIP. -mark On Jan 24, 2005, at 8:52 PM, Steven P. Donegan wrote: I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and gatewaying traffic from other countries will be considered to be 'theft' by the local governments/telco's (of their long distance revenues). My experiences were all with USA- Asian manufacturing locations - if you did voice-over-network between the same companies offices no sweat - the moment you allowed hop-off/hop-on gatewaying you were at risk to lose your phone lines!!! Samuel Tardieu wrote: Ed == Ed Guy [EMAIL PROTECTED] writes: Ed Now, Jeff Pulver has created Bellster(tm) - Half Napster/Half Ed Party Line - to fully realize the original vision. We've just Ed finished our testing and it is now open for your use. We'd love to Ed hear your feedback. This is awesome! I just setup my Asterisk server today, this could not be a better timing for me :) Feel free to call France landline numbers, my Asterisk is waiting for you :) Sam ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Codec negotiation
The codec is selected by asterisk depending upon the codecs that you have allowed for the particular channel context and your setting of the bandwidth= parameter. It would be nice if you could set things up so that an inbound call could force * to a higher bandwidth codec when needed (for example, an inbound fax call, let's say) but AFAIK this is not possible. -mark On Jan 25, 2005, at 10:18 AM, [EMAIL PROTECTED] wrote: Hello On every Incoming SIP and IAX call I see the following in asterisk debug: Accepting AUTHENTICATED call from 10.10.10.10, requested format = gsm, requested prefs = (), actual format = g729, my prefs = (g729|gsm|g723|g726|ulaw|alaw) priority = mine The problem is that the codec preference on both parties is different The calling party has preference gsm/g729/etc The called party (the one you see this debug from) has preference g729/gsm/etc The problem is.. This call is now set up with G729... And I want it rather to be decided by the callING party (thus want the call to be negotiated GSM) What can I do about this? (I just want that if I receive a call the calling party decides the codec, and not my side) My IAX.conf and SIP.conf have the following allow settings now Allow=all Allow=g729 Allow=gsm Allow=ulaw Allow=alaw Help :-) ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] IAXTEL is dead/dying?
As someone that's just recently setup an * server I agree. I thought about setting up an Iaxtel account as well but couldn't see the point in it because I had setup FWD for testing. I continue to use FWD for all my toll free calls and the quality is just awesome. I can't see how Iaxtel would provide any additional benefit. Perhaps the time for Iaxtel has come and gone. There are plenty of IAX2 providers these days, * has become quite popular, so the need for a separate telecom network doesn't make a whole lot of sense; not that FWD isn't separate, it's just more popular IMHO. -mark On Jan 21, 2005, at 6:12 PM, Michael Graves wrote: Yeah, FWD has been pretty good about their beta of the IAX2 support. My * server has been on it for 6 months without too much trouble. I even use it to bridge out to Signate.co.uk where my boss has an account. It was crystal clear last night from Houston TX to Cambridge UK. Dead reliable. I'm dropping my IAXTel registration when next I get around to such things. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 Inbound Sound Quality
Try changing to a less-bandwidth intensive codec (like GSM) and see what happens. -mark On Jan 21, 2005, at 7:08 PM, Brian Dingman wrote: I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't as crisp and clear as when I make outgoing calls over IAX. The other party does not hear any audio defects. Anybody have any suggestions on tweaking this? Or has anyone experienced the like? Running * 1.0.3 on an AMD 1700 with 512 MB of RAM (Red Hat 9). I am the only user currently on the system. I am connecting with their IAX server using ULAW and my SIP phone is also using ULAW (Sipura 2000). Thanks, Brian ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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: IAX Inbound Sound Quality
On Jan 22, 2005, at 10:49 PM, Michael Graves wrote: I notice that all four of my IAX2 based termination providers send incomming calls in trunking mode. You can tells since the command IAX2 Show Registry reports all the connections to port 8617. This is something that is determined at their end. In trunk mode I beleive that the jitter buffer is not effective. IIRC the jitter buffer is currently broken in trunk mode and should be turned off. http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 An alternative for testing is to set trunk=no in iax.conf. I've had to do that for my VPC trunks because I've also found that outbound faxing seems to be broken with trunking turned on (at least to VPC). FWIW, I had similar problems with VPC so I switched to Sixtel.net. No such problems anymore. VPC must still be using quite a lot of custom code or routing their calls in some weird way because I've found two problems with them so far while using IAX2: 1) The fax problem mentioned above. 2) Inbound DTMF is quite broken. (They are working on a fix and said it would be at least 30 days...but then in December they said it would take 2 weeks...). What a drag. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Short DTMF Tones and Asterisk
Have you tried calling into * using another phone or phone system? Try it from a cell. Try it from the pub, etc. etc. It may have nothing to do with the length of the DTMF at all because IAX2 sends DTMF out of band. FWIW, inbound DTMF is not working properly with Voicepulse Connect either right now when using IAX2 (digits are missed), but everything seems to work fine over FWD via IAX2 (at least the last time I checked). -mark On Jan 24, 2005, at 5:31 AM, Robert P. McKenzie wrote: I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE = CALLUK.COM 0870 = IAX Inbound The phone quality of the spoken call is fine, but DTFM tones aren't working. I'm using ulaw as the codec and bandwidth has been set to high in iax.conf. Any advice would be great. I could post debug logs of a call if someone would care to explain exactly what to capture. I'm still a newbie to Asterisk. Thanks in advance. -- Robert P. McKenzie | GammaRay Technical Services Ltd [EMAIL PROTECTED] | [EMAIL PROTECTED] http://www.uk-experience.com | http://www.gammaray-tech.com Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014 ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Damn DTMF Beeps on my calls
It's a common problem with VOIP and tends to happen when certain voices hit tones that mirror a DTMF soundwave. Some CPE's may be more sensitive and therefore more likely to cause problems in this regard. -mark On Jan 24, 2005, at 2:38 PM, Me wrote: Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] LiveVoip DTMF Issues
Same problem I'm having with VP Connect. Perhaps it's a question of the version of Asterisk being run. I'm on 1.0.2. -mark On Jan 24, 2005, at 11:36 AM, Brian Dingman wrote: I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at all. Is there anything I can do on my end to fix this problem, or is the old axim you get what you pay for true? It should also be noted that I have some other DID's from other providers and DTMF recognition is pretty much dead on. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Is Voice Pulse Connect good ?
Unless something's changed overnight Vonage doesn't offer a service that can be used directly with Asterisk; instead, you need to use an analog adapter and something that will give you an analog FXO port (like a TDM100). With Voicepulse Connect, VoipJet, and others, you don't need any extra hardware and you won't have any extra D/A A/D conversions. -mark On Jan 24, 2005, at 12:18 PM, Manjit Riat wrote: Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] LiveVoip DTMF Issues
Maybe you could contact the Voicepulse people since they're looking into it too. It seems to be a problem whether or not trunking is turned on or off. -mark On Jan 24, 2005, at 4:29 PM, Brandon Patterson wrote: Our people are looking at this right now and have been for the past few days. Use Asterisk Ver. 1.0.3 some people encounter it where others do not. * Problem is under investigation. Brandon Patterson LiveVoip LLC Same problem I'm having with VP Connect. Perhaps it's a question of the version of Asterisk being run. I'm on 1.0.2. -mark On Jan 24, 2005, at 11:36 AM, Brian Dingman wrote: I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at all. Is there anything I can do on my end to fix this problem, or is the old axim you get what you pay for true? It should also be noted that I have some other DID's from other providers and DTMF recognition is pretty much dead on. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] IAX2 trunking, Voicepulse Connect, and Outbound Faxing
I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if I have trunk=yes defined along with g.711 (ulaw), I am not able to send faxes out. I get a poor line condition, comms error, etc. If I turn trunking off the fax machines establish a connection quite readily and the fax goes through lickety split. So how is it that with trunking on I can receive faxes but not send? Perhaps this is a Voicepulse bug? And for anyone else wondering about Sixtel (iax.cc) faxing (at least outound) just doesn't seem to work at all through them. When a call connects you a hear a brief CNG tone and then the line goes silent. Maybe they have T.38 turned on. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 Recommendations Please
Just joined the list. Yay. Seems to me that thanks to VOIP people are now trying to do big things with still relatively low bandwidth connectivity and/or consumer-grade hardware. Little Netopia routers aren't going to get you where you need to be. The solution is to upgrade to either a Linux-based router (as Michael suggests) or to a bigger router hardware (not a cheap way to go about it though). -mark On Jan 19, 2005, at 9:05 AM, Michael Graves wrote: On Tue, 18 Jan 2005 20:01:51 -0600, [EMAIL PROTECTED] wrote: Hello all, We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will increase to 4/4 next year. The Netopia simply breaks out our WAN IPs, and we've got a switch hooked up to it on the inside (Actually I've got a QoS box in-between). - | Internet | | on Cat5 | - | - | Netopia | - | - | QoS Bridge | -- | -- | switch for WAN IPs | -- | | | -- | LAN Switch | -- | | | | Any recommendations on something that isn't as pricey as Cisco? I'm in discussions about us building Linux units down the line, but for now we need something we can buy. Cisco is too expensive for us. m0n0wall is my favorite. web site http://m0n0.ch/wall/ Review on Tom's Networking http://www.tomsnetworking.com/Reviews-161-ProdID-MONOWALL.php Open source running on either a plain vanilla PC or a Soekris embedded platform. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] something between an ATA and a channel bank for a small office?
Your upstream bandwidth is far too small. Remember, a T-1 (1.5Mbps/symmetrical) is used in a channelized setup to provide 24 64kbps telephone lines (so to speak). Trying to stuff too many calls into 256kbps using a low bandwidth codec is highly optimistic. Definitely not something I would do in a business setup. One of the problems with these residential grade broadband services is that users anticipate that the bandwidth will ALWAYS be available to them and that is just not how these services are designed. -mark On Jan 19, 2005, at 9:02 AM, Michael Graves wrote: On Tue, 18 Jan 2005 22:11:55 +, nik martin wrote: I have had very bad experiences with IAXYs so far.. I have pulled them and will be attempting a refund shortly. Bad audio, overheating and shutting down until allowed to cool, etc. make it unusable in a business environment. That said, is there a low-mid priced solution for a remote office to connect to a home office runing asterisk? There seems to be a hole in the market for 6-8 person remote offices. SIP isn't really an option because the remote office is fairly low bandwidth (1.4 mb down, 256k up ADSL). It seems my options are: 1. Inexpensive SIP phones connected to a local asterisk server which connects to my * server at the main office. 2. POTS phones + Asterisk + channel bank + t-1 card at remote office, connected to my asterisk server at main office 3. POTS phones and multiple FXS cards in * server at remote office with local T-1 line to terminate calls + IAX2 connection to main office for inter-office calls. None of these seem ideal due to the complexity of having a remote * asterisk server in the loop. It seems to me that your data rate is abitrarily low. If you could get a higher oubound data rate you'd have better results on average. I switchedmy home office ADLS from 1.5M/384k to 3.0M/768k and the impact was huge. There certainly are mid-market SIP phones available. I'm enamoured of my Polycom IP600, but the IP300 500 are less than $200 each. Avoiding FXOs entirely will eliminate a major headache. I think that the availability of business class features on SIP phones is worth more than the slight cost savings you might achieve using ATAs with analog phones. Polycom IP300 = $140 Polycom IP500 = $210 I'd be inclined to build an embedded Asterisk for the remote location and trunk back to your main office. You can also setup an account with an ITSP like Sixtel.net or Voipjet and place outgoing calls directly from the remote server. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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
