Re: [asterisk-users] Asterisk + iaxmodem + hylafax makes sometimes wedged for hylafax
On 07/04/2013 08:49 AM, Gianni Fioretta wrote: Hi, we have a faxserver with Asterisk, IAXModem and Hylafax. Faxes come from a SIP trunk to Asterisk, then are forwarded throught 5 IAXModems managed with Hylafax. Hylafax users can also send faxes to these modems and Asterisk send them throught the SIP trunk. We also have a dedicated modem used only for sending faxes coming from an Hylafax dedicated user. Sometimes Hylafax reports that a modem is wedged and this modem remain wedged until we restart IAXModem daemon. When all modems becames wedged the server can't send and receive any fax. IAXmodems do wedge sometimes, and I currently advise users to develop a resetmodem (see 'man wedged') to automatically reset the iaxmodem. It's not clear to me why the iaxmodems sometimes wedge. It essentially means that somewhere the iaxmodem code is either stuck in an endless loop, blocking on a read somewhere that shouldn't block, or something of that nature. If you could capture the gdb or strace of an iaxmodem as it gets wedged, then that would be most-helpful. However, I can tell you that the wedging occurs more-frequently in cases where there are other problems such as line audio quality issues or lots of non-fax numbers being used accidentally. So your use of SIP (VoIP) for what should be a lossless data channel is probably a factor there. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Configuration
On 11/05/2012 04:18 PM, Roy Abshire wrote: What is the best way for me to setup Fax Capability with VOIP only. You can use T.38, perhaps, if your VoIP provider supports it and you can get it working. But unless you need faxes to go through the telephony system (i.e. you have fax machines hooked up to FXS ports) I'd recommend using an on-line fax service such as Mainpine's instead of trying to do fax over VoIP. Fax over internet-strewn SIP generally isn't going to work very well. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. So on your MFP you'll scan it instead of using the system's fax capability, and then fax it through the online service. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/04/2012 09:27 AM, Carlos Alvarez wrote: However I'd just suggest that you look at the business case for screwing around with fax at all. As a society, if we had decided to stop supporting this dead technology years ago, with all the time and money we've collectively wasted we could have completely eliminated world hunger. I recognize that you're being a bit facetious in this latter comment, but the argument that you're making here is unfounded. I believe that if you were to look at the Davidson Consulting reports about the fax industry for as long as those reports have been available you'd find this. The technology is not dead and has enough momentum to propel it forward for many years to come. Maybe this is understood in your acknowledgement of society supporting it, but the reason why it's supported is because the technology is sound and fills a very valuable purpose in business and other activities. There is no adequate replacement for fax. E-mail doesn't do it, and most other reliable document communication mechanisms are locked-up in proprietary patents and interests that will invariably prevent them from becoming standardized at all. I'm not a big T.38 fan-boy, although I do applaud the ITU for that attempt to get fax working on IP networks. Unfortunately, it's fundamentally flawed because it needlessly perpetuates the tether between fax and telephony. In an IP network there is no reason whatsoever for fax to be saddled on top of a telephony layer. Fax is data communication, and IP networks are quite effective at data communication. I can envision a future fax system which truly uses modern IP network designs such as DNS, encryption, security, and rides on a very effective communication protocol and yet continues to operate on the fundamental communication protocol defined in ITU T.30 which makes well-implemented faxing so dependable. I can't count the hundreds of hours I've wasted on fax support just to prop up this stupid and unnecessary technology. Many others have felt exactly the same way, and I don't mean to be rude, but invariably the reason why they feel this way is because they repeatedly tried to do it the wrong way. We just made the decision this week to outsource it all and never deal with it on our network again. I am slowly re-gaining my sanity because of that decision. And until the new technology comes along that is and will be *precisely* the right decision for most of the people who move to a virtual environment or who completely detach themselves directly from the PSTN. Now I'm going to take a fax machine out to the parking lot and shoot it, even talking about this awful waste of time makes my blood boil. Well, if you were using stand-alone fax machines then that was part of your problem. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM Fax
On 08/17/2012 04:58 AM, Steve Underwood wrote: On 08/17/2012 06:08 AM, Eric Wieling wrote: Has anyone experimented with increasing the DAHDI chunk size in improve fax reliability? If so, did it help, hurt, or not make any difference? I haven't found issues related to the DAHDI chunk size. The main thing which used to hurt FAXing with Asterisk before Digium launched their own FAX software was the timing within Asterisk, which they refused to fix at that time (although independent patches were available). With the launch of FFA they changed chan_dahdi so on a FAX call the buffering should change to make the flow of transmitted audio a lot more elastic. People just tolerate some hiccups in voice calls, but hate latency. Modem signals must be rigidly timed, but a bit more latency is OK. This change fixed the main issue affecting all the FAX solutions around. If that switch in the buffering mode is not happening on your system for some reason it can badly affect the reliability of FAXes. I'm uncertain of exactly to which changes you're referring. Your comments seem to fall in-line with the notion behind the DAHDI buffers feature for the channel as well as the DAHDI fax-detection faxbuffers feature, but I'm seeing no noticeable improvement, AND I'm uncertain how to implement the CHANNEL(buffers) feature due to: -- Executing [4628160@fax-outbound:1] Set(IAX2/ttyIAX99-584, CHANNEL(buffers)=12,half) in new stack [Aug 18 20:12:40] WARNING[6381]: func_channel.c:530 func_channel_write_real: Unknown or unavailable item requested: 'buffers' -- Executing [4628160@fax-outbound:2] Goto(IAX2/ttyIAX99-584, outbound,4628160,1) in new stack -- Goto (outbound,4628160,1) -- Executing [4628160@outbound:1] Dial(IAX2/ttyIAX99-584, DAHDI/g0/4628160) in new stack On some installations there are occasional instances in most outbound calls where Asterisk creates what otherwise would be considered jitter on the DAHDI channel. Generally these do not cause much real-world trouble, but I'm a stickler for perfect audio quality on all-digital calls. I've seen this on Asterisk versions 1.4, 1.6, and 1.8. On other installations there never is any such trouble noticeable. Would you mind being a bit more specific on the Asterisk changes to which you refer and how they should be implemented in the configuration? Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.12 and Fax?
On 07/23/2012 09:23 AM, motty.cruz wrote: Hello, I'm trying get fax working over VOIP lines. I'm running Asterisk 1.8.12 server, working fine, however I would like to get rid of our anolog fax lines and integrate with our fax to our Asterisk Server. Any recomendataion from this list? I had done some research but nothing solid. Don't try to send faxes over LAN/WAN-strung VoIP channels. If you want to get rid of your analog lines, that's fine; use an on-line fax service provider. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.30 Fax session error: Received bad response to DCS or training
On 07/01/2012 01:46 AM, Petros Moisiadis wrote: Hello, On http://pastebin.com/AP5GBWUR you may see an excerpt from asterisk full log that includes a failing fax sending session. As you can see in line 328, the transmission fails with error Received bad response to DCS or training. It seems that something goes wrong along the lines 315 to 320, but I can't figure it out. Perhaps somebody with enough experience with the T.30 protocol can understand what is happening. What does this DCN 'ff 13 fa' response mean? I would highly appreciate any tip on how to get more info about this error or workaround it. I' m using asterisk 1.8.11.1, as does the other end. DCN is disconnect. So DCN in response to DCS means that the receiver didn't like something about DCS, TCF or possibly TSI so much that it decided to abort the fax by disconnecting after announcing it. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
On 06/07/2012 06:18 AM, Daniel Seagraves wrote: On Jun 6, 2012, at 10:47 PM, Lee Howard wrote: Unless you're going to move to an internet fax service provider you'll probably not want to attempt to switch your fax line to a VoIP line and still attempt to fax over it. And even then, depending on how much fax traffic you have moving to an internet fax service provider may not save you any money. I thought that was what iaxmodem was for? Part of the plan here was to dodge buying serial cards or modem banks when we started faxing more. IAXmodem was developed in order to make the modem hardware-agnostic. In other words, iaxmodem functions independently from the hardware, but that doesn't mean that you can provide a suitable audio channel to IAXmodem without some kind of hardware. Every once in a great while someone will come onto this mailing list or any number of others and announce that they've successfully got fax-over-VoIP going for them. In almost all of those situations that I've been permitted to analyse they're simply getting lucky in that ECM (error correction) is saving them, the remote senders and receivers support a well-implemented ECM, and the jitter isn't too bad where ECM couldn't remedy things. And I fully expect that at some point down the road things the VoIP provider will do things differently or the user will change some things on their network, and suddenly what used to work tolerably well for them will suddenly stop working. Save yourself this headache. So it doesn't matter what hardware you use: Digium, Sangoma, OpenVOX, etc. But you'll need to use some kind of hardware to interface with your PSTN service for fax. If your VoIP provider supports T.38 and if their T.38 implementation works with the T.38 implementation in t38modem or Asterisk 10 then you may be able to utilize one of those and avoid continued use of your PSTN fax connection. Understand, however, that almost all fax failures that you may have after that change will not likely be able to be resolved on your end alone. Nearly all fax protocol problems will have to be resolved by your T.38 provider, and depending on your relationship with them and the demand that you may have on high-reliability faxing this could be frustrating. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
On 06/06/2012 12:40 PM, Daniel Seagraves wrote: The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Unless you're going to move to an internet fax service provider you'll probably not want to attempt to switch your fax line to a VoIP line and still attempt to fax over it. And even then, depending on how much fax traffic you have moving to an internet fax service provider may not save you any money. my budget for this project is exactly $0. I can't afford to buy new devices. Unless your boss wants you to do VoIP from a headset on the PC I think you're chasing a lost cause. And, for what it's worth, $100 per month for two analog PSTN lines is rather typical. Depending on how much voice traffic you have and how much of it is local or inbound... switching to a VoIP service may not actually be a cost-cutting measure. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problem on direct FXO port
On 05/18/2012 04:45 AM, Sebastian Gutierrez wrote: with FFA I may get 70% of faxes ok. Nobody that I work with would consider that acceptable. Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
On 05/17/2012 07:40 AM, gincantalupo wrote: That could seem counter-intuitive but it is not. Not to mention the fact that information technology is not science, the solution to broken faxes is to lower down speed. The DSP algorithms change slightly between bitrates and considerably between modulations. Changing the modulation/demodulation algorithms can many times avoid problems caused by some type of line audio disturbance. This is why fax machines and fax applications are programmed to try various bitrates and modulations when training is failing at the default settings. Because historically most fax machines have supported V.17 14400 bps as a default it became customary for support technicians to suggest slowing it down to 9600 bps. (I think that the real intent here was to switch to V.29 9600 bps, but in practice it often results in V.17 9600 bps.) The purpose in this isn't really so that the communication takes longer (you can imagine that stretching-out data over a longer period would increase the likelihood of some audio disturbance affecting the demodulation), but instead I believe that the purpose in this recommendation is to cause a change in the DSP algorithms. Now, disabling ECM (error correction) is just plain wrong as long as the ECM protocol is implemented properly on both ends. If ECM protocol is implemented properly on both ends (and most are implemented well-enough that this applies to them) then ECM should be left enabled. By disabling a well-implemented ECM feature you're essentially making the claim that the remote-side ECM protocol is broken. If disabling ECM actually makes things work and you never get a corrupted fax image come through after that then it only means that one or both of the endpoints had faulty ECM protocol. Some technicians (including those working for fax machine manufacturers) will recommend disabling ECM if faxes aren't getting through. While this may have originated with the purpose of avoiding problems in faulty ECM protocol I think that any regular use of this suggestion is simply to get the customer to go away. The customer will see a page come through with streaks and lines, but it will be successful, and so they'll unfortunately be happy with that enough to let the technician off-the-hook with the disable ECM advice instead of actually fixing the real problem (either getting the line audio quality problem corrected or fixing the broken ECM protocol). My idea was to tell iaxmodem not to accept fast speed rates so the fax machine on the other side should be forced to negotiate a slower speed And what you were doing with the HylaFAX modem config file for the iaxmodem should have worked to do this. Why it wasn't working can only be determined by investigating your installation. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
On 05/17/2012 07:53 AM, Andrew Furey wrote: we use ActiveFax for sending (interfaced from an ERP package) and often get Comm Error 283 and incomplete faxes. If it's just making a bad situation worse, how is it that our solution of turning off ECM mode fixes it 98% of the time? I'm curious. Because apparently the ECM protocol in ActiveFax is broken. If disabling a feature that is designed to *improve* fax reliability and performance actually does the opposite, then there's no other explanation than to conclude that the implementation of that feature is broken in the product you're using. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax .pdf from Asterisk
On 05/03/2012 01:28 PM, Bruce B wrote: I want to send out 1000 faxes. I have an excel sheet of numbers and I have Asterisk 1.8 installed from repository. I don't want to use a fax machine or any ATAs or analogue equipment. How would Asterisk help me with faxing these? and what add-ons do I need to make this possible? Not interested in HylaFAX with IAXmodems? (I presume that you are using PSTN circuits and not VoIP.) Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5
The error happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. Alessio wrote: If I install asterisk i have the same problem. can anyone help me? thanks -- From: Lee Howard fax...@howardsilvan.com Sent: Thursday, September 01, 2011 6:29 PM To: ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore HylaFAX) should run on the same system as Asterisk. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes suddenly failing
kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes suddenly failing
Steve Underwood wrote: On 09/01/2011 11:50 PM, Lee Howard wrote: kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Because its the industry standard last resort of anyone who doesn't understand FAX and is using T.38. Even HP recommends for their own fax machines it numerous times: http://h2.www2.hp.com/bizsupport/TechSupport/Document.jsp?lang=encc=ustaskId=110prodSeriesId=378056prodTypeId=18972objectID=c00062808 http://h2.www2.hp.com/bizsupport/TechSupport/Document.jsp?objectID=buu02549lang=encc=uscontentType=SupportFAQprodSeriesId=3366988prodTypeId=15179 Yes, always a last-ditch effort, and if it actually succeeds in getting a legible document through then it means that either 1) the ECM protocol on either the sender or the receiver is gravely flawed, or 2) something that requires ECM (like V.34-Fax/SuperG3) ended up being disabled along with ECM and that the problem really had to do with that something and not with ECM. I've never seen a fax document that couldn't make it through with ECM enabled be able to come through legibly with ECM disabled otherwise. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5
Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore HylaFAX) should run on the same system as Asterisk. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Issues
Ryan McGuire wrote: Unless your network is under load and you are seeing dropped packets and high jitter, I would absolutely not do T.38. The cheapest and easiest approach that I have found is to buy yourself an FXS gateway and just make sure you are using ulaw. As SIP is usually running over UDP/IP it doesn't take much to produce dropped packets. Dropped packets mean lost audio which means lost data and possible demodulation difficulties for the modems. If you're in an environment where dropped UDP packets don't occur you're in a very rare scenario. For the most part people who claim success when faxing over SIP G.711 are being rescued by ECM (error correction) within the fax protocol. There are very, very few who really have mitigated UDP packet loss. That said, all T.38 systems are not equal. Certainly, the reliability of your T.38 provider may not be any better than that of G.711 fax over the SIP UDP. I only recommend faxing over TDM everything else is at your own risk. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Issues
Steve Totaro wrote: On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard fax...@howardsilvan.com wrote: Ryan McGuire wrote: Unless your network is under load and you are seeing dropped packets and high jitter, I would absolutely not do T.38. The cheapest and easiest approach that I have found is to buy yourself an FXS gateway and just make sure you are using ulaw. As SIP is usually running over UDP/IP it doesn't take much to produce dropped packets. Dropped packets mean lost audio which means lost data and possible demodulation difficulties for the modems. If you're in an environment where dropped UDP packets don't occur you're in a very rare scenario. I suppose you are talking about from the provider and not on the LAN? You certainly can (and usually will) have UDP packet loss on an uncontrolled LAN. At Equinix in Ashburn VA, I have never had a dropped packet via the crossconnect from our cage to Level3's cage. Sub ms pings. Putting the primary PBX in Equinix and a 100meg speed for all VoIP calls in our out. 100meg DIA and 100meg layer 2 fiber to corporate. I have no reason to doubt your claims, but if this is true, then your arrangement there clearly mitigates the likelihood of UDP packet loss. Nevertheless, this arrangement is not something that the typical user who asks how do I fax over SIP/VoIP is going to have. Without being very clear about the environment and explaining the pitfalls of not following your example exactly, you're not doing them any favors by encouraging them to attempt it in their environment. For every one user who I've ever heard from saying that they have reliable G.711 faxing over their SIP channels I've heard from a dozen who don't. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
Here are a few guidelines that I think may serve you well... Firstly, every network port that is being listened-to on any publicly-reachable system MUST be carefully protected - typically by firewalling. So, for example, you're likely going to want to block SSH from all but certain IPs. In certain situations you may need to expose a port to the entire world. In these cases you really have to take measures to limit the amount of probing that you allow from the entire world. One approach that has worked for me with SIP are these with iptables: iptables -N SIP_CHECK iptables -A INPUT -p udp --dport 5060 -m state --state NEW -j SIP_CHECK iptables -A SIP_CHECK -m recent --set --name SIP iptables -A SIP_CHECK -m recent --update --seconds 180 --hitcount 5 --name SIP -j DROP This rate-limits any source to 5 new SIP communication attempts every 3 minutes. If you service a lot of SIP devices all running behind one IP, then it may simply be wise to dodge this security by accepting all SIP communication from that IP... if that one IP remains static, of course. (I can't take credit for this... I found it shared on-line by someone else.) Secondly, disable the guest account in your sip.conf (allowguest=no). I recognize that this is enabled by default for the sake of convenience, but it's a nasty pitfall for those who are unaware of it. Lastly, in sip.conf set alwaysauthreject = yes in order to avoid revealing to a brute-force attacker when they have hit on a valid username. I'm sure there are many other good habits to follow that others here could share, but those come to mind with respect to the problem you've experienced. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive faxes
David Backeberg wrote: On Wed, May 4, 2011 at 12:00 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: (For my part, I'm actually surprised that nobody came up with a proper protocol for encapsulating the stream of zeros and ones that make up a fax transmission but rely on the precise timing inherent with a circuit-switched network, into something more suitable for sending over a packet-switched network. That would have fixed it good and proper.) They did. It's called TCP / IP. It allows sending PDFs, and they can even be encrypted. Faxing is for people who haven't heard of the internet. Nobody has said that faxing couldn't use TCP/IP... and there's no reason why T.38 couldn't use TCP/IP. Nobody has said that faxing couldn't use HTTP as a transport... or SSL... or any other kind of sensible mechanism. Why in the world people try to keep faxing (data transfer) tied-down to audio channels by putting T.38 into H.323 or UDP/IP SIP beats me. Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alarm POTS lines
Jeff LaCoursiere wrote: I don't think I have a prayer of hacking iaxmodem to do what is needed to emulate a modem though :) I can't pretend to know what an alarm system needs out of a modem, but as far as iaxmodem acquiring data-modem capabilities that part is already developing. IAXmodem inherits its DSP capabilities from spandsp, and you can see on the spandsp that certain modem types are already available in the newer spandsp snapshots. That doesn't mean that you can expect iaxmodem or spandsp to work for you anytime soon out-of-the box, but know that eventually you'll see this happen. Between the dockstar and the ATA (which I am already providing for dialtone anyway) it is around a $40 cost solution for me, and the customer gets to drop the POTS line, which is around a $30/month savings for them. I'd probably just eat the added cost to get the customer, and could even charge some modest fee for the alarm connection that would still in the end save them money. Look at fax, for example. Wouldn't we love to tell our customers to dump their old fax machines for scanners and email? Some people just won't until the thing catches fire or otherwise dies. There are many reasons besides ignorance that faxes (and thus fax machines) are still around. For one, there is a serious technological work-flow hurdle involved with the scan-to-email approach replacing fax completely because, for one, it's not like you can do that for every one of your would-be fax recipients. Consequently trying to replace faxing with a scan-to-email approach ultimately means that someone still has to do some faxing. As it's typically easier to send a fax than to scan/attach/email, work-flow productivity will actually drop by forcing the abandonment of fax machines (i.e. by utilizing a mail-to-fax service for intended recipients where a direct e-mail will not work). (I'm convinced that fax machines are with us for the long-haul. Now, whether or not futuristic fax machines operate directly with a POTS line or with IP connectivity seems clear that it will eventually become a hybridization, but I truly believe that those who engineer that future technology will necessarily have to divorce the IP-side of the systems away from the whole modulation/demodulation over audio channels bit. On an IP network it simply makes no sense to take a data stream, and modulate it to audio only to demodulate it back to a data stream again because the IP network makes the modulation/demodulation superfluous where it was necessary on the PSTN. So this running of T.38 FoIP over SIP VoIP is consequently a passing phase; it's not a solid-enough solution to replace traditional faxing.) So is there anyone out there with the DSP skills to do the iaxmodem-like part of what I describe above? Would a bounty raise any interest? As I said before, I think the work is already being done. And for what it's worth, the monthly $30 or $40 summed over years is a petty amount compared to the necessary development cost of the kinds of technologies you're discussing. You'll gladly pay $40 monthly for life than need to pay $100K to develop this stuff. A little more searching today turned up this: http://www.gouloum.fr/code/sm/sm.html Which is REALLY close to what I need... And note that it uses spandsp (albeit a newer snapshot than iaxmodem currently uses), so iaxmodem is therefore not far-off from where you claim to need it to be. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending T.38 fax negotiation problem
Kevin P. Fleming wrote: On 05/04/2010 06:30 AM, Miguel Amez wrote: I'm experiencing the same problem with t38modem and hylafax. My problem is that on the re-Invite phase it syncs lower than 2400 bpps and the connection hangs on the second page. The patch I'm talking about won't affect t38modem and Hylafax usage at all. If the re-INVITE arrives before you have connected the call to t38modem, the negotiation process will very likely fail. Typically HylaFAX users have the calls connected to the modems from the outset. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Foip solution
Mike Diehl wrote: I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? It's reliable as long as you're not using FoIP (i.e. as long as you're faxing with PSTN lines). Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detected digit 'f'
Kingsley Tart wrote: DEBUG[18902] chan_dahdi.c: Detected digit 'f' This happens just after the initial fax negotiation has started and seems to correspond with the sending fax machine giving up. Turn off fax detection. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem / hylafax receive problem
Kingsley Tart wrote: Jan 14 12:44:49.39: [ 3403]: -- [9:AT+FRH=3\r] Jan 14 12:44:56.39: [ 3403]: -- [0:] Jan 14 12:44:56.39: [ 3403]: MODEM Empty line Jan 14 12:44:56.39: [ 3403]: MODEM TIMEOUT: waiting for v.21 carrier Jan 14 12:44:56.39: [ 3403]: -- data [1] Jan 14 12:44:56.39: [ 3403]: -- [2:OK] iaxmodem cannot hear any fax signaling in the call. Jan 14 12:44:56.39: [ 3403]: -- [9:AT+FRS=7\r] Jan 14 12:45:26.39: [ 3403]: MODEM TIMEOUT: reading line from modem Jan 14 12:45:26.39: [ 3403]: MODEM Timeout Jan 14 12:45:26.39: [ 3403]: Failure to receive silence (synchronization failure). Jan 14 12:45:26.39: [ 3403]: -- data [1] Jan 14 12:45:26.41: [ 3403]: -- [2:OK] However, there is *some* kind of audio on the call. It would seem that this test call is producing some kind of long-duration bad audio sounds which are not detectable by the modem as fax. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Faxing
Miguel Molina wrote: Please correct me if I'm wrong, but AFAIK spandsp based fax applications for asterisk only support a maximum of 9600bps. No. V.17 (speeds up to 14400 bps) are supported. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] allowguest defaults to yes for SIP
In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote: On Thursday 12 November 2009 07:47:34 Lee Howard wrote: In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Actually, they only have access to your default context. Whether you make available outgoing calls in your default context is your choice. By default, there is no capability of making outgoing calls from your default context. Well, yes, the default configuration is useless. But, let's say I follow doc/security.txt exactly and have this: [default] exten = 6123,Dial(Zap/1) ... therefore, by default, an unauthenticated user from anywhere can call the extension Zap/1. It's not my point whether or not this poses a financial risk. My point is that this is an insecure default behavior to have allowguest=yes. Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. The purpose of the allowguest option is to allow persons to call into your system from a zero-knowledge position. This allows you to publish a general SIP address as a point of contact. These people should need to deliberately use allowguest=yes. I would venture to guess that these people already know who they are and deliberately have this set. I would venture to guess that there are far, far more people who have it turned on by default who really don't want it that way than there are who expected it to be that way and desire it to so be. The reason why it is set that way in the sample configuration is to make it easy for new users to get to that magic moment when Asterisk first responds to their call (in essence, to get the user hooked). This is a poor excuse for a poor default security setting. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Correct, you should be using something like internal instead. And yet this point is not even made clear in the doc/security.txt file. It says to not use default for anything you don't want to get abused, but it doesn't say *why*. So I can envision, then, someone reading the document and then changing context=internal in the [general] section of sip.conf... and thinking that they responded correctly to what the document said. If this default is to persist then I think that it behooves the developers to at least make this exposure clear to the users. Therefore, the in the [general] section of sip.conf the context should not be set to default, but rather to unauthorized or public or open or free or something that makes it clear that this is where unauthenticated SIP calls go. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote:
On Thursday 12 November 2009 09:53:17 Lee Howard wrote:
These people should need to deliberately use allowguest=yes. I would
venture to guess that these people already know who they are and
deliberately have this set. I would venture to guess that there are
far, far more people who have it turned on by default who really don't
want it that way than there are who expected it to be that way and
desire it to so be.
And the people who use this probably believe that YOU should be the one
who has to deliberately turn this option off. I would venture to guess that
90% of all statistics are made up on the spot, including this one and the
two you specified above.
I made it clear that they were guesses. But, please *DO* take a vote on
this. I'm not seeing anyone but you stand up to support the default
setting. Unless you take a vote there's really nothing I can do but guess.
The fact that this problem is being exploited leads me to believe that
this is far-more prevalent a problem than just my single case. Take
care of your users when you can do something so easily. Don't
deliberately let them learn things the hard way on the basis that they
should have known better. The mere fact that this issue is addressed in
doc/security.txt should be an indication that there is a common risk
that could be averted.
And yet this point is not even made clear in the doc/security.txt file.
It says to not use default for anything you don't want to get abused,
but it doesn't say *why*. So I can envision, then, someone reading the
document and then changing context=internal in the [general] section of
sip.conf... and thinking that they responded correctly to what the
document said.
You've just made a case for enhancing the documentation, not for changing
the defaults. If you contribute documentation changes to this effect on the
issue tracker, I would be more than happy to commit them.
The patch is attached. Feel free to add it to bug tracker issue ID
16226 which some maintainer was happy enough to close already.
And, for what it's worth let me restate my vote that the default for
allowguest be changed to no on the basis of keeping ignorant people
from making a stupid mistake.
Thanks,
Lee.
--- asterisk-1.4.21.2/doc/security.txt.old 2009-11-12 09:53:03.0 -0800
+++ asterisk-1.4.21.2/doc/security.txt 2009-11-12 09:56:38.0 -0800
@@ -48,12 +48,15 @@
Therefore, you should NOT allow access to outgoing or toll services in
contexts that are accessible (especially without a password) from incoming
-channels, be they IAX channels, FX or other trunks, or even untrusted
-stations within you network. In particular, never ever put outgoing toll
-services in the default context. To make things easier, you can include
-the default context within other private contexts by using:
+channels, be they IAX channels, SIP channels, FX or other trunks, or even
+untrusted stations within you network. Keep in mind that the default setting
+for SIP configuration is allowguest=yes. So unauthenticated SIP users will,
+by default, be able to access the context specified in the [general] section.
+Therefore, never ever put outgoing toll services in the public context.
+To make things easier, you can include the default context within other
+private contexts by using:
- include = default
+ include = public
in the appropriate section. A well designed PBX might look like this:
@@ -63,9 +66,9 @@
[local]
exten = _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
-include = default
+include = public
-[default]
+[public]
exten = 6123,Dial(Zap/1)
--- asterisk-1.4.21.2/configs/sip.conf.sample.old 2009-11-12 09:57:19.0 -0800
+++ asterisk-1.4.21.2/configs/sip.conf.sample 2009-11-12 09:58:41.0 -0800
@@ -24,7 +24,7 @@
;
[general]
-context=default ; Default context for incoming calls
+context=public ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote: The issue in question was suspended, while the reporter makes the case on the Asterisk-dev mailing list, which is not this list. The opinions there amongst contributors (meritocracy, not democracy) are that keeping the sample configuration as it is now is probably the way to go. Sigh... of course. It's a gentlemen's club and only members have a say. If you want to create a new issue and attach your patch there, I'll look at it. I sent a patch. I pointed you at a case. That should have been FAR more than enough for my attempt at contribution to be acceptable. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Danny Nicholas wrote: Gentlemens clubs usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. You're suggesting that if I pay someone they'll be able to get the default setting for allowguest changed to no ? I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. So either I am stupid or lazy or there is a risk here that can catch even others off-guard. I've been down this contribution road-path a half-dozen times before with Asterisk. So forgive me if I don't play it out to the final futile note. In ESR's CatB there's the idea where the maintainer encourages (and wants) bug reporting, feedback, and other non-code forms of contribution (as well as code contributions). He refers to it as grooming co-developers. That's not how Asterisk development works... here you can contribute if you're already in the meritocracy, but if you're not, then you have more than a difficult time in trying to even contribute in small non-monetary ways. So anyway, I've been down this road a half-dozen times already, and it ends up being futile, frustrating, and time-consuming. I'm too busy today to be interested in playing. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Michiel van Baak wrote: When I started working with asterisk, and found my first issue, I created a patch, put it on the tracker, followed up on the comments, and stuff got in. I'm sincerely pleased to know that you've had a different experience than have I. If you read the page about contributing code to asterisk, it clearly states that the dev mailinglist is the place to discuss development. If you post comments there, people will read it, comment on it, and if more people agree with the ideas it will get implemented. It's how all OpenSource projects work. I truly wish it were. I've seen more than a few that didn't. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Michael Wyres wrote: The way I see it, the reason you have encountered some resistance to your opinion in regards to whether guest access should be allowed by default or should not be, is not because your opinion is right or wrong - everyone is entitled to an opinion - and your stance has merit, certainly - I don't think anyone is actually disputing that. It is more that a lot of the people on this list have been using Asterisk for a LNG time, and have explained why it might be advantageous to have guest access enabled by default. There are definitely uses for this functionality, as has been demonstrated by a number of examples contained in this thread. I certainly understand why someone would want such a feature. Again, I think that it's a feature that should not be enabled by default. I realize that some people that are using this feature would be inconvenienced if this default were to change. I think that inconvenience is far-outweighed by the benefits in avoiding exploitation who are unaware of this feature. I don't know how long, exactly, a LNG time is. Certainly there are plenty of people who have used it longer than I have. I started investigating and studying Asterisk in 1999. I started using it in 2002. If that's not long enough to deserve a voice, then I understand. Isn't this why you joined the list? To learn more about the product, and get ideas and assistance from the more experienced users of the product? I've been a list member for a very long time. Back in that day I was accustomed to joining the users list for every software I used with any interest. The point of joining the list was, yes, to learn, but also to share and to provide feedback to developers. You raised your concern, and Tilghman (a senior developer at Digium) explained the reasoning behind the default setting. He suggested that you take your concern to the tracker and post a patch. You resisted. In case you weren't aware, I *DID* open a case on the bug tracker, and I *DID* write a patch as requested. However, an eager bug marshal decided to close my case before the patch was written and asked me to come to this list to discuss the subject. So Tilghman was asking me to create a *NEW* ticket and to post the patch there... yet all the while there were discussions going on on asterisk-dev about the very same subject which, as clearly stated, superseded my contribution due to merit. In other words, there was little point for me to write any patch until after those whose opinions count due to merit are done (but even then, I still wrote and contributed a patch). But please understand, I've been down this path before many times. I wasn't trying to be resistant. Instead, I was merely cognizant of the fact that I had already done enough to express my opinions and that to continue restating them over and over would have been futile and argumentative. Now, the default extensions.conf contains the following snippet: snip [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include = demo /snip Now, a lot of people never RTFM for anything. Moreover, how many people actually read the EULA for any piece of software they use? It's not Asterisk/Digium's fault if people don't read the available documentation that they provide. The quite plainly clear statement above is in a production system, you probably don't want to have the demo there. Did you read that bit? Did you wonder why that bit is there? Yes, I did read that. This led me to immediately remove the demo. It did not, however, lead me to set allowguest=no. When I first started working with Asterisk, I clearly remember that line (or something very similar) piquing my curiousity to dig a little deeper as to why that statement was made. Lo, I discovered that this was because by default, guest access is allowed. You certainly took it further than I did. I accepted what it said at face-value. I didn't continue to investigate. I can't help but think there are others like me who will not read between the lines to learn that guest access is enabled by default. Indeed, the language in doc/security.txt doesn't currently make this clear, either... reading it at face value I see a bias against using the default context for anything involving tolls, but it still doesn't say that unauthenticated callers are permitted by default. Again, you were more inquisitive than I was. I applaud you for it. Do we expect that level of inquisitiveness from all users? I too found the default access odd at first, but I chose to understand the reasoning from people who knew better, instead of chucking a hissy fit. I'm sorry, I'm not sure I understand your definition of hissy fit. If you view my behavior as a hissy fit then I do apologize. Please understand, however, that I *DID*
Re: [asterisk-users] Asterisk 1.4 and Fax
I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. Lee. Dan Journo wrote: I've heard mixed reports. Some say they've had no problems, some say that faxes fail most of the time. I want to try it out and see how it goes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 02 November 2009 18:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) Does Asterisk 1.4 support T.38? Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing over Carrier SIP trunk/g711 ?
Jason Aarons (US) wrote: Anyone have a customer sending/receiving multi-page faxes over Verizon Business SIP trunk/g711 ? Verizon Business indicates they don’t support it, and I have 2 recent customers that it doesn’t work for, and 1 current large customer telling me he’s going to make it work grin. The issues is the latency/jitter on fax/g711 over Verizon Business seems to spit out only 11 pages of a 15 page fax. Anyone having faxing over PSTN SIP over G711 that is working? Any advice? Read: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Machine Testing ...
Gordon Henderson wrote: Not FAX over VoIP, but testing 2 FAX machines back to back ... The scenario is that a client has one of my asterisk VoIP only systems which they're happy with, but need to test FAX machines, so rather than plug in a TDM card with FXS ports, I'm suggesting something like a PAP2T device with 2 analogue ports registering to their asterisk box to use to send test faxes from one fax machine to the other ... So as far as asterisk is concerend, it's just 2 more SIP extensions and I'll arrange re-invites, so data ought to stay inside the ATA (if possible) rather then have asterisk handle the media (but even so, it's a lightly loaded switched ethernet LAN) Anyone see any issues with this, or suggest a better ATA than a PAP2T ? Sending faxes between two ports on the SPA-2102 works for me. However, there are other PSTN line simulators that will be less maintenance over the long-run. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] No Carrier detected sendig fax with Hylafax-Iaxmodem-Asterisk
Johann Steinwendtner wrote: Lee Howard schrieb: fred wrote: That’s being said, before going through the T38 Gateway tests, I’ve tried first the Fax2mail and Mail2fax solution with (Hylafax + Iaxmodem + Asterisk), to make a well-tested Asterisk solution working and I’m already facing some problems. Receiving faxes is ok but sending faxes gets stuck into “No Carrier Detected”. Debugging leads me to the following remarks: I) _Carrier or Called Station Identifier or (CED) tone_ from the called fax machine is received since recorded (iaxmdm-iax.wav attached) The audio coming from the receiver is quite audibly corrupted (just listen to it and then compare to when you call a fax machine over TDM). No doubt you are attempting to pass fax audio through VoIP. Please read: Is there a tool where I can feed the captured audio (e.g. extracted from tcpdump) in order to get a fax image ? Can hylafax in a certain mode do that ? I suspect that you're asking a different question from where the original poster was headed (hijacking the thread). So, assuming that we're talking about clean audio and not corrupt audio as the OP provided... There may be some utility with spandsp to do this. As HylaFAX is not itself a DSP there is no such capability in HylaFAX. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] No Carrier detected sendig fax with Hylafax-Iaxmodem-Asterisk
fred wrote: That’s being said, before going through the T38 Gateway tests, I’ve tried first the Fax2mail and Mail2fax solution with (Hylafax + Iaxmodem + Asterisk), to make a well-tested Asterisk solution working and I’m already facing some problems. Receiving faxes is ok but sending faxes gets stuck into “No Carrier Detected”. Debugging leads me to the following remarks: I) _Carrier or Called Station Identifier or (CED) tone_ from the called fax machine is received since recorded (iaxmdm-iax.wav attached) The audio coming from the receiver is quite audibly corrupted (just listen to it and then compare to when you call a fax machine over TDM). No doubt you are attempting to pass fax audio through VoIP. Please read: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax Driver
Tilghman Lesher wrote: What's the use case for the Digium driver? Am I missing something by not using it? While they accomplish the same goal, the commercial driver is based upon a different codebase, Ok. provides support for patented fax protocols, Really? V.34-fax (33,600 bps) is supported? I had understood differently. and enjoys far more testing than Steve can reasonably do for an unpaid side project I can't speak for how much testing the T.38 or T.30 sides to spandsp have had. However, the T.31 and V.17, V.29, V.27ter, and V.21 aspects have undergone a *tremendous* amount of testing and development scrutiny (as these are used in IAXmodem). I am aware of single installations that communicate successfully with very arbitrary kinds of fax machines in the USA which alone have communicated several millions of pages of fax in the last two years. If there ever is a problem (and it is extremely rare) I hear about it. So I don't know of what kind of testing you speak, but I would caution you to reserve judgment simply on the basis that it is an unpaid side project. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax Driver
Tilghman Lesher wrote: On Sunday 07 June 2009 19:39:50 Lee Howard wrote: Tilghman Lesher wrote: What's the use case for the Digium driver? Am I missing something by not using it? While they accomplish the same goal, the commercial driver is based upon a different codebase, Ok. provides support for patented fax protocols, Really? V.34-fax (33,600 bps) is supported? I had understood differently. I would research the patents involved, but I am prohibited by employment contract from exploring patents granted. Due to said employment contract prohibitions you can't tell me whether or not Digium's Fax Application supports V.34-fax (33,600 bps)? My understanding is that there are certain aspects of fax that are still under patent, Yes. Specifically V.34. If my understanding is correct the relevant patents expire in a few years. and those are provided (along with indemnification) by the commercial driver. Understood. But it was my understanding that V.34-fax was not supported by Digium's Fax Application. And if that's correct, then there are no patents for which indemnification is necessary. That's not to say that a commercial fax driver does not have its place with some customers. I only want to clear up any misrepresentations about possible patent infringements by spandsp to which you alluded. I'm not suggesting that the commercial driver is more reliable, only that it enjoys far more testing. Again, regardless of your knowledge of how much testing goes into your employer's product, I question your ability to know with any degree of certainty as to how much testing has been involved with competing products. I certainly know that *I* have no clue with regards to spandsp other than the testing to which I've been witness. So I am curious to know how you are able to make such assertions. That said, hours of use in production do not speak to the amount of testing done. Scrutiny of production use exposure does not constitute testing? Well, I would argue that you cannot possibly test real-world conditions without actually placing the test system into the real-world with real-world use (thus, production). I cannot think of a better way to test software than to eventually put it into real-world production use and then have the developers monitor those systems closely. IAXmodem is a completely different ball of wax, and I think you would agree that if the builtin FAX support in spandsp provided excellent support, there never would have been a reason for IAXmodem to be developed. I'm interested to know how you understand my intent in developing IAXmodem differs from what I recall. I developed IAXmodem because I needed to interface HylaFAX through an Asterisk PBX without purchasing additional hardware (other than the T1 cards that were already involved). Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing issues
Todd S wrote: Our call path is Sip trunk from MAX TNT - Asterisk - T1 - Adtran endpoint converting sip trunk to copper line for house wiring. Users at the endpoint can receive faxes without a problem. However, sending faxes are not so friendly. 1 out of 5 faxes will send successfully. The remaing 4 continually fail with the fax machines reporting Poor Line Quality. QoS is setup on both ends and working as expected across all traffic to and from both sides of the call. Voice quality is good on calls from the same lines. I've checked all sides and had all vendors look at their equipment and they are all pointing the finger at each other. I've got sip traces from both ends and both look good with the exception of random BYE's being sent from the Adtran side of the call during random fax calls. T.38 is enabled on both ends as well. Anyone have any possible ideas where to look further or an idea of how i could at least improve the situation? Is T.38 actually being used? If not, then please see: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf If T.38 is being used then it's probably due to a T.38 bug in one or more of the gateways, and you'll need to work with the provider of that T.38 gateway for a resolution. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium fax failing
Michael wrote: Anyway this is a great example of why MICROSOFT is worth billions, and Linux has to be given away. Not because Microsoft is L33T but because the majority of the stuff sold for Windows works out of the box. For what it's worth, their fax software doesn't work very well out of the box. :-) Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium fax failing
Michael wrote: On Mon, 27 Apr 2009 03:40:12 you wrote: Slightly off topic, but M$ is worth billions because they started in 1976 or so, became the de facto standard, and were pretty cutthroat in the way they do business. They have a profit motive and have always taken the path that makes them bigger with bigger profits, even to the point of fighting antitrust allegations. That creates billions. Linux is quite the opposite, and Asterisk is a friendly midpoint between profit and open source. YMMV. CF Almost every commercial customer simply want's something that works. Unfortunately, with my over a decades worth of experience with Linux, if I can't get T38 to function properly, it's a hack, not commercial grade software. It sounded to me like the T.38 functionality in Callweaver was working for you. Why did you abandon it? Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing and TIFF files
D Tucny wrote: 2009/4/22 Michael mich...@networkstuff.co.nz mailto:mich...@networkstuff.co.nz I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing. Does anyone know of a way, either while producing the file, or after, to tell how many pages have been produced? (without manually viewing the file) tiffinfo? then count the number of data blocks... HylaFAX's 'faxinfo' tool would also work. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3
Marco wrote: I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine. They are linked together through localhost. I've turned qualify on for the iax peer. Periodically I've this message: [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer: Peer 'iaxfax' is now UNREACHABLE! Time: 3 [Apr 20 23:47:56] NOTICE[4632]: chan_iax2.c:8128 socket_process: Peer 'iaxfax' is now REACHABLE! Time: 3 It happens a lot of times during the day, even when the box is not loaded at all. What does iaxmodem say? (Look at the iaxmodem logs.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax for Asterisk questions
Steve Underwood wrote: I wonder how much demand there is for colour FAXing. It's quite niche. When it's available people will have their fun with it for a few times, and after the novelty quickly wears off they never use it again. I have quietly enabled color receiving support on high-volume fax receivers, just to see what kinds of things would come through in color without advertising it in any way other than the DIS signal. The incidence of color fax reception was on the order of one in ten thousand - and in no cases did any of those faxes have meaningful color content; someone had simply pressed the Send Color button on their fax machine instead of Send Black. And, in fact, the data size of a JPEG fax image is so much greater than that of a monochrome fax image (especially JBIG) that you just can't leave color fax support enabled without seriously introducing some significant cost risks (channel usage) even for those one-in-ten-thousand cases. In other words, when people are done with the novelty in color faxing, and when people are aware of the actual cost involved with color faxing (even accidental cases) they will disable it. Thus, color faxing is really only ever used in very deliberate cases - where both the sender and the receiver have negotiated ahead of time that such a communication is being made. That said, color faxing *does* have its niche. For example, real estate appraisers will commonly have color photographs in their appraisal report, and they will deliberately ensure that the color aspect is not reduced to monochrome. Another good example is that of an advertising agency in returning proofs to its customers. In both of these cases e-mail attachment tends to be the dominant and preferred mechanism, but color faxing can - in some cases serve as an alternative (where the sender and receiver both have capable equipment). It actually requires some real work, and is not just a matter of linking to libjpeg. The colour space for a JPEG FAX image is different from the colour space used for most PC and camera JPEG images. That creates some real messiness. Correct. libjpeg can be used, but libjpeg natively knows nothing about the required ITULAB colorspace... which means that the application would need to perform the necessary transforms. That becomes a lot of work. HylaFAX+ has partial support for colour, but I think the colour spaces hassles mean it was never completed (I think it just works in one direction). Yes, it only supports color receive, and to do so it requires a specially-patched version of libjpeg (and libtiff). I've made a couple efforts to try to drum up enough interest in order to get libtiff capable of performing the JPEG colorspace transforms to and from ITULAB using a native libjpeg-6b, but so far I've always come up short. And, to complicate matters, finding someone who already knows enough about JPEG colorspaces and libjpeg who is available for hire is a tall order... which means that you've really got to be dedicated and interested in learning it yourself. However, the aforementioned patches to libjpeg and libtiff already *will* allow for conversion to and from ITULAB colorspace. However, the patches are messy and will never become accepted upstream. (Don't count on libjpeg ever releasing a new version any time soon, and the internal JPEG support in libtiff changed in a way that breaks the way that the patches worked.) So if someone really wanted to they could use them to learn from. They work with HylaFAX+ to receive only because when sending the image needs a tagline put at the top which I never got around to developing for color. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sequential Ring Groups?
Marshall Henderson wrote: However, when a fax is complete on that modem and another call comes into it, the modem is still in a state of 'settling down' from the last call and I'd like to have it ring a different channel if possible. iaxmodem stays busy for 5 seconds after going on-hook and for 5 seconds after receiving a reset command (ATZ). In my experience this is plenty of time to cover the settling down such that you won't need to even consider it. Just do the normal hunt group dialplan approach as suggested here: http://iaxmodem.sourceforge.net/faq.php Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
David Backeberg wrote: It may be possible to use hylafax, but I don't know how or why you would. The reason *why* is generally due to support issues. For one, HylaFAX probably has a better T.30 implementation in its Class 1 driver than does app_fax. At least that historically has been true. I welcome the day when all fax applications perform as well as HylaFAX has since its 4.2.x days. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
Steve Underwood wrote: Lee Howard wrote: David Backeberg wrote: It may be possible to use hylafax, but I don't know how or why you would. The reason *why* is generally due to support issues. For one, HylaFAX probably has a better T.30 implementation in its Class 1 driver than does app_fax. At least that historically has been true. I welcome the day when all fax applications perform as well as HylaFAX has since its 4.2.x days. It used to be true, but I suspect there is little in it when you use spandsp-0.0.6pre7 or later. :-) That's wonderful news. :-) Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX reliability
Hi Steve, Steve Underwood wrote: In chan_dahdi.c there is now code that extends the buffering inside dadhi when a FAX is detected, and puts the buffering back to normal at the end. This isn't really a cure - its more of a bandaid. However, I expect it has the desired effect if they have put it into the trunk code. You need to enable this feature in chan_dahdi.conf. Very interestingly, many people who have problems sending from app_txfax or app_fax have no problem sending from iaxmodem + HylaFAX. It seems Asterisk can feed a zaptel/dahdi channel much more reliably when passing a signal through, than when generating on in an app. So are the chan_dahdi.c developments mentioned of any practical value to iaxmodem users? Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX reliability
Steve Underwood wrote: Hi Lee, Lee Howard wrote: Hi Steve, Steve Underwood wrote: In chan_dahdi.c there is now code that extends the buffering inside dadhi when a FAX is detected, and puts the buffering back to normal at the end. This isn't really a cure - its more of a bandaid. However, I expect it has the desired effect if they have put it into the trunk code. You need to enable this feature in chan_dahdi.conf. Very interestingly, many people who have problems sending from app_txfax or app_fax have no problem sending from iaxmodem + HylaFAX. It seems Asterisk can feed a zaptel/dahdi channel much more reliably when passing a signal through, than when generating on in an app. So are the chan_dahdi.c developments mentioned of any practical value to iaxmodem users? That's a good question, and I have no idea about the answer. Some people who have had problems sending from app_txfax say iaxmodem + HylaFAX works OK on the same machine. This seems strange, as you might expect a problem in scheduling I/O would affect passthrough as well as applications. What I don't know is whether they just have a lot less trouble with iaxmodem, or they have no trouble at all. Do you get reports from people who say receive is stable, but transmit is not? There have been a few reports of that situation, but all of them that I recall had to do with people who were trying to use VoIP for fax (HylaFAX and iaxmodem tend to be much more tolerant of the audio cut-outs caused by jitter than are other receivers). Otherwise, no, I don't have any of those situations that I can point out. I suppose that I'd need to make some recordings to say for certain whether audio cut-outs were occurring. However, what would really be nice would be to see some comments come from whatever developer at Digium made those chan_dahdi.c modifications. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
Steve Totaro wrote: IAX2 has been a lemon since it's inception. Sure, some people have success. It seems to work OK for IAXModem. I chose to use IAX2 in developing IAXmodem because IAX2 is relatively simple compared to SIP and because at the time I didn't know of any easy-to-use SIP library with a simple sample program. libiax2 and its accompanying sample programs were ideal for getting things going quickly. In retrospect, I still think that it was a good decision. However, if I had chosen to use SIP instead it would make things like adding T.38 support into it much easier. (But, that said, my purpose in developing IAXmodem was to interface HylaFAX with TDM cards through Asterisk without additional hardware. T.38 support wasn't - and still isn't - ever a goal.) Yet, I did have to work hard to get libiax2 working right. I think the problem is that from the slightest version change to another, two boxes with different versions and many times the same version don't work well. The root cause of that, I suspect, is that Asterisk itself does not use libiax2 for chan_iax2. Consequently, developers really don't have a reference library that is used as the standard. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Marco Signorini wrote: Analyzing your answers, seems that fax handling is still today problematic with IAXModem and Hylafax... or I'm wrong? A single server that I administer, receiving 12,000 pages and sending 1,000 pages daily would seem to contradict your conclusions. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi wcb4xxp and fax
stoffell wrote: I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Make sure that you're using the latest mISDN drivers. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum version for asterisk and iaxmodem
James Lamanna wrote: Hi, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when asterisk gets a call from iaxmodem, it says that the format for the call is unknown. Does anyone know if there is a minimum version of asterisk that is compatible with iaxmodem 1.1.0? I originally developed IAXmodem while using Asterisk 1.2.x. I have since migrated to Asterisk 1.4.x. I never attempted to use Asterisk 1.0.x with IAXmodem, and I have also never tried Asterisk 1.6.x (although I suspect others have without issue). Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hylafax asterisk iaxmodem problem
voip crazy wrote: This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. What does the CLI say about this call? Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (announce) asterisk T.38 gateway
Matt Watson wrote: I'd probably be a little pissed if I were Steve Underwood if somebody pocketed over 10k $USD for taking credit for a product that my free library did the bulk of the work for. I can't speak for Steve at all, but any major contributor to an open-source project faces this, so maybe my comments can be helpful in understanding at least how *I* feel on this matter. As an open-source contributor I have tried to make it very clear to those who choose to use my contributions that *no* thanks or other expressions of gratitude are required or expected. I have already received my just reward; I make sure of that before I even undertake the development process. Certainly a thanks here or there is welcome and heart-warming. And best of all is when someone else chooses to help in the development process. But none of that is expected or required. I've worked for (and received) a bounty before, but in that case I did not release the code until after the bounty was received or in the hands of an escrow agent. As T.38 gateway development for Asterisk was worth so very much more than US$10K, I can't help but believe that Steve's motivation was from elsewhere, and that he did not necessarily have his sights on that goal. However, if it was *me* who had put up that bounty I would double-check that the claimant actually did the work before I paid out... and I would divide up the bounty payment accordingly if it was not all done by that person. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Jay R. Ashworth wrote: On Thu, Jun 12, 2008 at 08:02:24AM -0500, Tilghman Lesher wrote: One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. Your phrasing, here, Tilghman, suggests that you mean that the administrative account should be renamed from root to madams, and I'm fairly sure you don't actually mean that. You actually mean create a regular user, and lock the machine down so that's the only thing that can be used to log into it, at which point, when and If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. ...you can use sudo to get it. Never, ever, ever, expose sshd to the public internet without firewalling. Only let trusted IPs reach sshd. The risk of brute force success, however small, is still far too great. Again, do not expose sshd to the general public. And for that matter... it's generally unwise to expose any service to the general public when the general public has no business using that service. A little bit of time learning some iptables basics will go a long way here. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
Benny Amorsen wrote: Now, what's the story on Hylafax+? How is that different from Hylafax? See: http://hylafax.sourceforge.net/about.php Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
Gordon Henderson wrote: Something that's always confused me - IAXmodem is built on spandsp. RxFAX is also built on spandsp. RxFAX gets one step closer to the data stream (no copper/IP in the way), so why are people using ( suggesting) Hylafax over RxFAX? spandsp includes a lot of stuff. In spandsp there are the fax modems: V.21, V.27ter, V.29, and V.17. There is also a Class 1/1.0 DCE interface (T.31). There is also a T.30 (fax protocol) driver. There is also a T.38 driver in there. There is other stuff there, too. IAXmodem uses the fax modems and the T.31 DCE... its main purpose being to function with other applications that have their own Class 1 T.30 drivers such as HylaFAX. RxFAX (and TxFAX) also uses the fax modems, but doesn't use the T.31 DCE, but instead uses the T.30 driver. So the difference underneath the hood really has to do with where the fax protocol engine/driver is. When using IAXmodem the protocol is done by HylaFAX. When using RxFAX/TxFAX the protocol is done within spandsp. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
Sanjay Rajdev wrote: We are using Asterisk 1.4.13 on FC6 and have a T1 card with 20 DID's. We are wanting to use one of the DID's for Fax, is this possible or do we have to add some addition Hardware and what is the best way to do this. http://iaxmodem.sourceforge.net Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
Steve Totaro wrote: You may need an additional server just to handle faxes if you are running many instances as they are CPU intensive. iaxmodem is not CPU intensive. 100 of them aren't. You can put that many on a typical modern machine and have them all faxing simultaneously and not see a dent in CPU due to iaxmodem. Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Machine Options
Joseph L. Casale wrote: It turns out their SIP provider doesn't support the T.38 protocol for faxing. Their statement is if you really need it, use ulaw and AstraFax? I don't understand how AstraFax makes a difference in the process? It doesn't make a difference. uLaw over SIP/UDP still involves jitter which results in data loss ... regardless of what fax device or program you use. As I tried to indicate in my first reply... I would encourage you to order an analog phone line for that fax machine. There are other options, but in most cases the customer is happy to pay the line charge. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Machine Options
Benny Amorsen wrote: Lee Howard [EMAIL PROTECTED] writes: Note that if you have a fax machine that performs some variant of T.37 (fax-over-email) and you have an on-line service provider that is willing to work with you... then you can rather easily get your fax machine faxing through their service. (Which is yet another option.) Tell me of a fax machine which can be programmed to do T.37 while keeping the same UI as regular faxes, and we'll buy a hundred. All the ones I've seen require you to type in email addresses instead of phone numbers, or use a separate keyboard instead of the usual numeric one. That is not acceptable for regular users. Unfortunately, I'm not well-versed on all fax machine UI's out there. I know that several of them perform scan-to-email or T.37 variants of some kind... Panasonic was one of them. However, whether or not it was a seamless and transparent switch to T.37 from analog I am not sure... because I never really used them. In the case that I needed T.37 I just built my own fax machine. Realize that you, too, can build your own fax machine with a scanner, a PC, and a modem. And if you are just doing T.37 you can omit the modem part. I used SANE utilities with some PHP to make a simple UI that ran on the small PC. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Machine Options
Andreas van dem Helge wrote: Cisco gateway with T.38 support. That's the only real way to do faxing through asterisk. Although this statement has marginally more truth to it given the SIP-only context that the original poster provided, it is still substantially inaccurate. There are several ways to do T.38 other than with a Cisco gateway. Now, if you meant that T.38 is the only way in the SIP-only context (and not specifically Cisco-branded T.38) then that has significantly more accuracy to it. However, if by *real* you also mean *reliable* then be aware that T.38 over SIP/UDP has an inherent weakness due to the medium that make it, in my experience, significantly less-reliable than simply having a fax machine hooked up to a traditional analog line. When my clients come to me with the same issue I generally do not push them into a corner with T.38. In almost all cases they find that it is worth the $20-50 monthly for the analog fax line... and if that expense is too much then the on-line fax service provider is an easy recommendation. Note that if you have a fax machine that performs some variant of T.37 (fax-over-email) and you have an on-line service provider that is willing to work with you... then you can rather easily get your fax machine faxing through their service. (Which is yet another option.) Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a distro with hlyafax rolled in?
James Finstrom wrote: elastix *shivers* Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to establish handshaking with fax machine
mark morreny wrote: I am simulating the sending of fax using sendfax through voip Ooops. Please see: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXModem - NDID=s
Louwrens Benadé wrote:
NDID=s
What the hell!? Why is 'NDID=s'?
Probably because at the point when your dialplan sends the call to
iaxmodem ${EXTEN} is s.
Thanks,
Lee.
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Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Joshua Kinard wrote: So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my Rolm phone, I get several seconds of silence followed by error tone. But on asterisk's CLI, I see this: -- Starting simple switch on 'Zap/2-1' -- Starting simple switch on 'Zap/3-1' -- Starting simple switch on 'Zap/4-1' -- Starting simple switch on 'Zap/1-1' So, okay, there are four calls coming in on the Zap (strange, but...) -- Executing [EMAIL PROTECTED]:1] Dial(Zap/2-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s -- Call accepted by 127.0.0.1 (format ulaw) -- Format for call is ulaw -- IAX2/iaxmodem0-5 is ringing -- IAX2/iaxmodem0-5 answered Zap/2-1 iaxmodem0 correctly takes the first call... -- Executing [EMAIL PROTECTED]:1] Dial(Zap/3-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s [Feb 15 15:40:22] WARNING[24329]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem0-1' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/3-1' status is 'CHANUNAVAIL' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s [Feb 15 15:40:30] WARNING[24327]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem0-3' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/3-1' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s [Feb 15 15:40:35] WARNING[24327]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem0-4' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/1-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/4-1' And the other calls get busy and improperly run through the auto fallthrough process (you *need* a Hangup in your dialplan fax-in context). The Rolm gives me error tone just before the Starting simple switch messages begin to appear, so it's almost like the Rolm is not waiting around long enough for the asterisk server to answer, before it jumps to the next configured T1 channel, runs out of channels (I only configured four in the Rolm and on asterisk). I think that your zaptel/zapata configuration between the Rolm and Asterisk on that T1 is misconfigured. Set it up for PRI if you can... it'll be a lot easier, is my guess. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Joshua Kinard wrote:
Another attached text file shows what iaxmodem is doing during all of this.
Something about adjusting skew.
[2008-02-15 17:11:12] Adjusting skew to -50.
[2008-02-15 17:11:12] Adjusting skew to -100.
[2008-02-15 17:11:12] Adjusting skew to -150.
[2008-02-15 17:11:12] Adjusting skew to -200.
[2008-02-15 17:11:12] Adjusting skew to -250.
There is no mechanism for iaxmodem to pull clocking right from Asterisk
other than examining the IAX2 timestamps. So in the event that iaxmodem
isn't getting voice frames from Asterisk iaxmodem is left to use
clocking solely from the system clock... which may likely not be in-sync
with the T1 clocking... and so iaxmodem the skew messages you see is
an attempt by iaxmodem to compensate for a clock skew between the system
clock and the timestamps on the IAX2 frames... but because iaxmodem
isn't getting any voice frames you get a run of these skew messages
until the call disconnects.
Added, how does this look?
exten = s,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten = s,2,Busy
exten = s,3,Hangup
Better. :-)
Thanks,
Lee.
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Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file The reason that you need the nojitterbuffer in the iaxmodem config file is because you're actually getting at least some jitter. IAXmodem's jitterbuffer simply fills-in gaps due to jitter with previously-heard audio samples. There is no way to recreate the missing audio. Filling-in the gaps with previous audio samples is effective in preventing premature carrier loss conditions, but it messes up the modems until real carrier loss does occur. It turns out that in most cases it's better to simply skip over the missing audio. The DSP seems to handle that quite gracefully. Thanks, Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Al lists wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. I think that you mean to say HylaFAX and IAXmodem or txfax/rxfax ... because spandsp is but a DSP/DCE library, and it cannot work alone, and iaxmodem uses spandsp. Thanks, Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk hylafax iaxmodem
Giedrius Augys wrote: I have problems with asterisk and hylafax+ iaxmodem. I can successfully send faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have problems: No carrier. This is hylafax log, maybe you can suggest me where to find ... Oct 17 07:38:48.22: [22428]: -- [16:ATDT37052390906\r] Oct 17 07:39:30.86: [22428]: -- [10:NO CARRIER] This is when you need to use the iaxmodem record feature (see the README file or http://iaxmodem.sourceforge.net/howto.php) to make recordings of the call and to examine the iax recording to see why it thinks that there is no fax signalling from the receiver. Or... you just send those recordings to me (the *.raw files that appear in /tmp/). Thanks, Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma vs Digium: (was Re: ping too)
Matthew Fredrickson wrote: Not to ignite any fires, but I don't think I've *ever* knowingly received a patch to libpri or chan_zap from them. And I've fixed a few protocol related bugs in libpri for people with Sangoma cards. It'd be nice if they at the very least supported the protocol stacks and zaptel channel driver they use to make money off their cards. The report appears to have been reaped from Mantis, but I was involved with a contribution from OpenVOX for zaptel, and from my perspective it looked like the Digium staff involved killed it and never gave any indication that the contribution would be accepted. Certainly seeing that kind of antagonism isn't going to encourage competitors to contribute. There is an atmosphere of hostility between Digium and its competitors that you yourself are expressing in this very thread. Expecting those competitors to eagerly come to your table and play in your pool underneath your rules... and then complaining publicly against them when they don't is really a bit much. Any Digium competitor is immediately on unequal footing with respect to Asterisk due to the dual-license and requisite disclaiming of contributions. You're asking those competitors to contribute not only to the open-source Asterisk, but also to contribute to Digium's ABE and private licensing ambitions. In my estimation what you're complaining about is only fair-play. If you really want fairness then start by being fair yourself. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX detection not working
Joe Acquisto wrote: As I understand it, I must have faxdetect = incoming to enable detection of the fax tone. Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on. It's a fax extension in the context where the call is at... not a fax context in the dialplan. Lee. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
Christian Peter wrote: Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and works most time. *chuckle* Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] asterisk, iaxmodem, hylafax quality problem
Thomas Kenyon wrote: The weird thing is, looking at the motherboard manual for my test machine, The lower the Interrupt does not neccesarily mean the higher the priority. Eg. 8 to 15 have a higher priority than 3 to 7. Correct. IRQ 2 bridges to IRQ 8. Thus the priority order is: 0, 1, 2, 8, 9, 10, 11, 12, 13, 14, 15, 3, 4, 5, 6, 7 This is one reason why on modern Linux kernels where the ATA (IDE hard drive) driver is permitted to be very resource-greedy the serial ports on IRQs 3 and 4 can lose requisite attention for high-throughput serial devices (like Class 2.1 fax modems). And just think of those poor, poor printers on the LPT port, IRQ 7... The end-result is that the already slim pickings on IRQs gets reduced even further to a very narrow band for add-on PCI devices, usually just 9, 10, and 11 on many systems. This is one reason for APIC, although it's quite buggy in many kernels and motherboard BIOSes. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] asterisk, iaxmodem, hylafax quality problem
Thomas Kenyon wrote: My zttest results weren't quite as bad as the previous poster. Home Machine. --- Results after 113 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.994452 This should be perfectly fine. Work Machine. --- Results after 115 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.993920 This should work fine as well. Are these results good enough to be able to use TxFax/RxFax/iaxmodem? I can't really speak for TxFax/RxFax, but they should be fine for iaxmodem. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general
Faris Raouf wrote: The problem is that when I run faxstat, it does not show hylafax connected to any tty. You're probably not running faxgetty (and your later comments below confirm this...) And when I try to run faxaddmodem (just to see what might happen) and select ttyIAX, I get an error saying that hylafax can't detect the speed of the device and that I should set it manually, and then iaxmodem promptly crashes at that point. You're probably not using HylaFAX+. The hylafax.org releases kill iaxmodem when faxaddmodem is run. See: http://hylafax.sourceforge.net/ Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to find t38modem
The site lists openh323.org, but that's the old t38modem I think. The new one should be coming out here sometime: http://www.voxgratia.org/downloads.html However, right now I think that the SIP-compatible version is only in CVS: cvs -z9 -d :pserver:[EMAIL PROTECTED]:/cvsroot/openh323 co ptlib_unix cvs -z9 -d :pserver:[EMAIL PROTECTED]:/cvsroot/openh323 co opal cvs -z9 -d :pserver:[EMAIL PROTECTED]:/cvsroot/openh323 co t38modem Lee. Gunnar Schaller wrote: Same site, just a few lines later: ... you could run Asterisk and Hylafax with T38modem (by www.openh323.org) on the same box and terminate T.38 calls ... Gunnar Hello, From http://www.voip-info.org/wiki/view/Asterisk+fax you can read: *Update Jul 2007:* For a T.38 gateway you can use Asterisk 1.4's T.38pass-through support in combination with the new OPAL (Open Phone Abstraction Library) - using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Where can you find this t38modem stuff ? Google replies things that doesn't seem to match (T.38-SIP termination). Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax
marek cervenka wrote: hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is changed in 1.4 in g711 pass-through faxing? thanks Jitterbuffer behavior, maybe? Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk
Michael Rice wrote: Hello. We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3 When we send faxes the people who receive the faxes complain that they look wrinkled or smashed up. Sometimes they are missing random lines. Has anyone seen this happen, or know how to fix it? Well, firstly, this is probably best handled on the iaxmodem lists, but... How are you interfacing with the PSTN? Zaptel, I presume? iaxmodem and Asterisk are on the same box? What does 'cat /proc/interrupts' say? Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] software bloat - is this really useful to anyone?
http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any plans for proper faxing support
Andrew Joakimsen wrote: I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. You can use Asterisk 1.4's T.38 pass-through support in combination with the new OPAL-using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax and Asterisk
Andrew Nowrot wrote: I am trying to build reliable fax solution with asterisk, iaxmodem and hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3 1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After installing the newest zaptel and wanpipe-3.1.0 beta I did zttest and it didn't give me good results: 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% Are you having trouble with fax? Rumor is it that the Sangoma hardware isn't as needy this way as is the Diguim. I'm not sure about that, though. In any case, what does your /proc/interrupts file say? My guess is that your A101 is coming after something, like your hard drive interface or your LAN interface or something. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Throughput
Don Kelly wrote: I've tried timing faxes two ways: From a fax machine on a station port of an AltiGen PC/PBX served by an MCI PRI calling back into the same PRI and reaching a RightFax server on a station port behind the AltiGen. From the same fax machine on the same station port of the AltiGen PC/PBX served by the same MCI PRI calling a number on an XO PRI connected to an Asterisk system (Digium TE410P), dialing out on another channel on the same PRI back into the MCI PRI and reaching the RightFax server on the station port behind the AltiGen. extensions.conf includes: exten = 6122353002,1,dial(zap/g1/6122590773) Sending a one-page fax with moderate density (no graphics) takes almost five minutes longer going through the Asterisk server. The longer transmit time is probably a result either (or both)... 1) retransmissions due to the audio being consistently corrupted, and ECM retransmissions to correct the corruption 2) training failure (probably due to corrupt audio) resulting in a slower transmission rate (e.g. 9600 bps vs 14400 bps) As to how to fix it... it's almost certainly audio degredation occurring in your Asterisk configuration or linkage... so debug your Asterisk setup. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Bottom line on fax reception
Steve Totaro wrote: Please qualify your usage. A couple faxes a day, a couple hundred, a couple thousand, or a couple hundred thousand? Couple hundred thousand per month - at least on one installation. Are you running asterisk and hylafax on the same machine? What is your TDM connectivity? Yes, same machine, TDM is PRI, usually... at least it is on the installation I am mentioning. Hylafax uses quite a lot of CPU juice. Huh? Certainly much, much less than Asterisk. Anyone ever scale up a quad T1/E1 server for faxing using asterisk and hylafax? Must be a heck of a server! It's okay. :-) Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Gordon Henderson wrote: The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. If you needed this you would handle the fax/voice detection in Asterisk, and only route the call to the iaxmodem if it detected fax. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Luki wrote: So essentially you need a pool of iaxmodems running on different ports, and then Dial() them until you find one that accepts your call. Or did I get that wrong? That seems really like a drawback to me The biggest drawback with app_rxfax is that if it crashes for whatever reason (happens sometimes), it will take down the entire PBX and all sessions with it. So you'd rather have the entire PBX crash in order to avoid creating sufficient iaxmodem instances to handle your fax call load? Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Luki wrote: So you are saying a pool if iaxmodems and a loop through Dial() to find an open one is the way to go? Yes. You could launch iaxmodems on-demand - as well as the corresponding faxgetty and wait for the initialization to finish - but it is better to just have a bunch of modems ready and waiting. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX on PRI and TE205P
And 10,545 downloads in 33 releases. Crazy people! nik600 wrote: its in a beta state with only one member... is it a stable project? thanks On 4/23/07, Lee Howard [EMAIL PROTECTED] wrote: nik600 wrote: i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? I would suggest that using IAXmodem with HylaFAX would be more stable than using an ATA-connected HylaFAX modem. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX on PRI and TE205P
There you go, it's no longer beta. That should solve all of your problems. Should you need testimonials there are plenty of them in the asterisk-users archives, on various wikis around, and you can surely get some from the iaxmodem-users list as well. Lee. nik600 wrote: sorry, i absolutely don't wont to minimize this project, i've just noticed that it is in a beta state, and i need a stable solutions, for a business activity. Can you or someone else give me some feedback? I know that Fax over Voip doesn't yet have a stable and complete support, but i have to use it in a LAN, where the most important Voip problems are minimized... thanks On 4/23/07, Lee Howard [EMAIL PROTECTED] wrote: And 10,545 downloads in 33 releases. Crazy people! nik600 wrote: its in a beta state with only one member... is it a stable project? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX on PRI and TE205P
nik600 wrote: i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? I would suggest that using IAXmodem with HylaFAX would be more stable than using an ATA-connected HylaFAX modem. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Blast over IP?
Wiley Siler wrote: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? No, but I can recommend that you read this to see why you shouldn't bother: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Blast over IP?
Wiley Siler wrote: Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? My suggestions are in the reading material. Basically it boils down to you not using VoIP for fax. Lee. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX thru TDM400p
Joe Acquisto wrote: AFAIK, the FAX targets are normal FAX machines, on the PSTN. What happens is, there appears to be a dial out, and a FAX negotiation, but it always fails. What does zttest say about your zap card configuration/installation? If it's not always 99.98% or better then it's due to hardware resource constriction and you need to escalate the zaptel card's priority on the hardware (like putting it at a lower IRQ). Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX thru TDM400p
Joe Acquisto wrote: Lee Howard [EMAIL PROTECTED] Wrote: 4/6/2007 2:15 PM: Joe Acquisto wrote: AFAIK, the FAX targets are normal FAX machines, on the PSTN. What happens is, there appears to be a dial out, and a FAX negotiation, but What it always fails. What does zttest say about your zap card configuration/installation? If it's not always 99.98% or better then it's due to hardware resource constriction and you need to escalate the zaptel card's priority on the hardware (like putting it at a lower IRQ). Lee. zttest does not exist on this system, Suse 10 based. IIRC, I never found the file(s) needed to compile it. It's usually built and left in the zaptel source directory where you extracted and built zaptel. If it doesn't get built for you from zttest.c then check the Makefile that it has zttest in BINS like this from mine: BINS=ztcfg torisatool makefw ztmonitor ztspeed zttest fxotune Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX mISDN
LKS GMAIL wrote: Does anybody know how to receive send faxes throw mISDN? It's almost impossible! I know that IAXmodem users are doing it. They have to get the right version of the mISDN stuff, though, I think. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Faxs any help :)
younss azzayani wrote: and this is the /var/spool/hylafax/log/c1: http://pastebin.ca/403282 cat /var/spool/hylafax/log/c3 :: http://pastebin.ca/403291 What does zttest say? If it's below 99.98% then hardware configuration is where the problem is. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
