Re: [asterisk-users] AstriCon videos: a question of method (Robin)
Robin, Thanks for the viddler.com suggestion! I'm uploading all of the ClueCon videos to it right now. John, so far I'd have to give viddler.com two thumbs up. I'm adding my stuff here: http://www.viddler.com/explore/cluecon Your ClueCon presentation should show up some time on Friday. I've noticed that there's a little bit of a lag time between upload and video being available for viewing but that's completely reasonable under the circumstances. Let us know what you decide. -MC ___ -- 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] John Todd, Moises Silva Speaking At ClueCon 2009
Hi Folks, I just wanted to share with you all some information about two well-respected members of the OSS telephony community who will both be speaking this year at ClueCon http://www.cluecon.com. Their topics are relevant to Asterisk users so I felt compelled to let everyone know about them. First, John Todd is going to be speaking on the subject Open Source Telephony In An Economic Downturn. I think we can all appreciate that topic. :) Secondly, Moises Silva is going to be speaking about his experience in developing modules for Asterisk and FreeSWITCH. Anyone who has subscribed to this list for any length of time knows that Moises is a great developer and a wonderful support of OSS telephony. We look forward to his insights. More information on this here http://cluecon.com/node/28. Thanks for your time. Hope to see many of you in Chicago this summer! -MC P.S. - ClueCon is technically a developers conference, however we discuss topics that impact users of all kinds. If you consider yourself a regular Asterisk user and you have an idea for something that you feel developers should be discussing then please email me off list. ___ -- 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Date: Fri, 21 Nov 2008 16:20:28 -0600 From: Terry Wilson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER in my case) was if I needed to load balance calls across multiple asterisk servers. If most of the phones are not separated by a NAT from Asterisk (as would be the case in something like a University network), the registration timeout could be set to a relatively high value w/o causing any problems which would cut down on some of the SIP traffic from registrations. In fact, I just ran some tests using SIPp and w/o any audio, using realtime w/ 10k accounts I can register 100/second while doing 10 calls/second. If you are looking just at registrations every 15 minutes or so, that is 90k devices that could register to asterisk. This was using 1.6.0.1 on my little HP amd64 development box--not anything near the kind of machine that you would probably install in a large installation. Asterisk just gets faster and faster. Some of the it isn't good at x stuff comes from experiences with older releases. In a HA and/or high volume scenario I worry about stuff like this that has been in tree since 1.0 or earlier and is in 1.6, channel.c lines 3825~3828: /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ That's not something I want in my high-end, high-capacity, high-availability production system! For smallish installations, this probably isn't a big deal given today's hardware capabilities. Still, it makes me wonder what other gremlins are out there that might bite me in a big-time install. At least with OSS I can see stuff like this. I shudder to think what psycho spaghetti code is running on Cisco, Avaya, Nortel, NEC, Shoretel, etc. -MC If you are lucky enough to have a situation where you can re-invite media and keep it off of the asterisk box, it can handle huge loads. Terry ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
If you would point me, i would gladly take a look at this patent list, for now my searches were unsuccessful. The ITU maintains a list of IPR (Intellectual Property Rights) claims for various technologies. Check it out: http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS On the left-hand side there's a search box, plus you can select G.729 (or one of the many derivatives thereof) from the recommendations drop-down list. When I select G.729 and click Search I get back a list of 52 items, most of which seem to be patents that have at least one claim related to this codec. (I see lots of references to stuff like CS-ACELP and other super-geekish acronyms that only smart people like Steve Underwood actually understand!:) IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Happy researching, MC ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
I wonder if they've got patents on various strains of Anthrax... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, October 01, 2008 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1) IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Happy researching, MC I keep a stash of 1,000 500mg sipro. Gotta be prepared these days -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
[asterisk-users] Question: Soft phone for ACD agents?
To those running call centers I have a question: what kinds of soft phones, if any, do you use? I'm wondering what is out there that has some hooks for custom applications or host system integration, etc. OTOH, do you prefer a desk phone for any reason? If so, why? Thanks for your thoughts, Michael ___ -- 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] FTC Bans Prerecorded Telemarketing Drivel
Gives us legitimate telemarketers a bad damn name. :-) Isn't legitimate telemarketers an oxymoron? -MC ___ -- 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] PRI TBCT - Practical Experience, Anybody?
You should expect that; in fact, that's what the 'TB' in 'TBCT' stands for... for a time, there are two B-channels involved. TBCT is a method of taking two existing already connected B-channels and linking them together into the network, it is not a 'transfer' facility where you provide a target DN and an existing call is 'transferred' to that destination. That feature is ELT (Explicit Line Transfer) and may also be known by other names, or possibly Call Deflection (CD) depending on whether you do it before the call is answered or after. In the scenario you outlined, the original caller (party A) calls this mediator (who answers as party B1). They then place a call (party B2) to you (party C), which you answer. Once that call is established, they can TBCT party A and party C, thus dropping the party B1/B2 legs. You will never see party A's identifying information on the call to you unless party B decides to provide it to you in some fashion; the network signaling would never know to provide it to you, since this is not a call transfer in the RDNIS sense of 'call transfer'. -- Kevin P. Fleming Kevin, This answer is excellent! Very well-worded and definitely useful to help someone grasp the idea behind TBCT. -MC ___ -- 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] Purchasing Digium IVR Prompts.
Try GM Voices. $6.95 per prompt plus $175 studio setup fee. To make it truly cost effective it might be worth it to find other users who need prompts recorded and then you can split the setup fee. Even if you have dozens or hundreds of prompts the fee is what is. I think they charge a separate fee for each voice talent so if you need prompts in different languages you'll have a setup fee for each language. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, July 29, 2008 10:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Purchasing Digium IVR Prompts. Just went to order some IVR prompts from the digium web site From the digium web site: We have created an easy and cost effective way to have customized recordings done quickly and with no hassle. I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single order, but then you'd need to split up the prompts yourself with audio software. That is NOT hassle free. 2. Since prompts are recorded seperately, each shows up in the shopping cart as a separate item. There is no way to see what the requested prompt is! We're going to have a lot of these (remember, each prompt is different), and keeping track of them NOT hassle free. 3. From the web site Also, you have the ability to upload your own intonation file to ensure a personalized and professional recording every time. what the heck is an intonation file? Is it a text file? Is it an audio recording? What format? The web site doesn't seem to say. Lack of documentation on the web site is NOT hassle free. 4. Of course, when I called customer service, they had no clue. NOT hassle free. Doug. ___ -- 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] Building an IVR
Hmm... You may be in one of those positions where there just isn't a great solution because your environment has so many constraints. You might want to check out the way freeswitch handles IVRs, dialplan hooks, FAGI-ish connections, etc. It will still take some work, of course, because there isn't an out-of-box solution (that I'm aware of) that can meet all of your requirements without lots of time/money/effort. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Monday, July 07, 2008 10:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Building an IVR So, I need to build a complicated IVR with Asterisk, with a lot of back end hooks. The dial plan itself has a lot of limitations, not the least of which is that the dial plan is ugly, hard to maintain, and full of gotchas like all variables being global etc etc. I've been involved with Asterisk for a couple of years now and this is a problem I have yet to see a good solution for. 1. I looked at VXML but it has too many integration problems. 2. AGI has overhead. 3. Fast AGI has single point of failure problems (we're using Asterisk 1.2 which bombs out the call when an AGI request fails), and has too many moving parts for what should be something fairly simple. 4. I'm aware of res_perl, but am not a fan of the maintainability of perl. 5. I looked for a valid link to res_python, but couldn't find anything. 6. Adhearsion? Looked at it a few months ago but couldn't work it out. There was too much 'voodoo' going on. 7. I'm not a C programmer, so writing a custom module, is both overkill and not feasible. Do I have any other options? Doug. ___ -- 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] Trouble with PRI config
Agreed. It looks like you've tried to tell the Avaya to be the network side but it doesn't seem to be acting like the network. Do what Steve suggested and see if you get a different result... -MC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config pri_net usually when connecting to a legacy system. Thanks, Steve T ___ -- 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] Trouble with PRI config
You'll probably need to turn on pri debugging for this span and then capture the output from when you connect the T1 cable. That might yield some clues, like whether or not any activity is happening on the d-channel and if so, if there are any errors that might tell you what's going on. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eve-Ellen Cole Sent: Thursday, June 19, 2008 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Trouble with PRI config Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] Using a Loopback Plug for an RJ-45 EthernetInterface for testing a Digium Card
I can't speak to exactly what the alarm status stuff does if the port you're looping expects to have a PRI plugged into it: I would expect Green, but no actual traffic, but I could be wrong, I'm a bit new on that front. Just for the record, this is generally a correct statement. I can't speak for every T1 interface out there, but every T1 interface I've personally used does respond to the so-called hard loopback that is described elsewhere in this thread. The hard loopback really is just a layer one test. On Zaptel you should definitely see a green light when you plug in a loopback test plug. This is kind of a sanity test - not necessarily for the equipment but rather for the guy trying to make it run! :) It is a very basic test, and it fits in with is the computer plugged in and turned and did the zaptel driver(s) get loaded and such. A green light simply means that the layer one connection is working - the physical connection, the framing and coding, timing, etc. Obviously this needs to be working before you can get meaningful traffic moving over the interface... Happy T1/E1-ing. -MC ___ -- 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 3rd party developed commercialsoftware sales licensing platform
Ok, I''ll bite. The question is: Do we want asterisk to contain a licensing engine ? That depends on the implementation. Your questions, I'm sure, will be discussed on the call tomorrow. Such an engine would need to : Hand out license tokens to proprietary modules linked to asterisk (like codecs etc) Hand out license tokens to proprietary systems connected to asterisk via manager (HUDs, etc) Hand out license tokens to proprietary endpoints talking to asterisk (softphones, media-gateways etc) The other question is this: does Asterisk itself *need* to contain the engine, or does it simply need to be available in case it's required for a specific 3rd party app(s)? That leads to another point for potential discussion: can the engine be self-contained and generic enough to the point that it is a utility that can be extended and used with other OSS, and maybe even proprietary, software? (Yes, there are GPL issues to think about, but assume for a moment that there are ways around any GPL issues and then think about the question.) One benefit to having a 3rd party commercial software sales licensing platform to work with Asterisk (or anything else) is that it allows for standardization. If a lot of 3rd party developers used a standard licensing platform then that would cut down on user confusion as well as the drain on resources that Tim mentioned. Anyway, that's just another $0.02 from Mikey. -MC ___ -- 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] RE:Asterisk 3rd party developed commercial software sales licensing platform
Gentlemen, Dean Collins alerted me to this thread which I had skipped over. (Thanks, Dean.) I thought I'd offer my viewpoint on the matter; please take it for what it is - just another opinion, although I hope it is an informed one. From my personal experience with buying software, licensing, and even music online, I've come to the conclusion that the best way to monetize an application or module is to make it easy for your paying customers to pay. Since thieves and hackers will always find ways around any security it is pointless to spend lots of time and money making something uncrackable, especially if that security implementation becomes onerous for your paying customers. My viewpoint is this: make it easier to do a legit install than to circumvent the security and you'll get most paying customers to pay. Thieves don't generate revenue but paying customers do, so do your best to make it easy for them to pay. That's my two cents, anyway. I'm definitely interested in other viewpoints, contrary or otherwise. This discussion is definitely an important one for OSS. -Michael ___ -- 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] Good article about VoIP, etc.
Gang, I know some of you like to keep up-to-date on various VoIP-ish happenings. Here's an interesting little article about FreeSWITCH that also mentions Asterisk: http://digg.com/software/Freeswitch_Poised_to_Shake_Up_the_Open_Source_V oIP_Scene The author guesstimates that Asterisk has roughly 95% of the OSS telephony market. I'd be interested to know if anyone has hard facts about the market share that Asterisk/Digium enjoy, both from the OSS telephony perspective as well as compared to the big commercial vendors. -MC ___ -- 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] Developer Conference, Aug 5-7, Chicago
Question: is anyone planning on going to the Cluecon convention this year? (www.cluecon.com http://www.cluecon.com/ ) I'm hoping to go this year and I'm hoping to meet other OSS telephony users and developers. BTW, Anthony Minessale said that there is a need for Asterisk speakers, so if you're an Asterisk user (or expert) and you're in the Chicago area in early August then perhaps you could check out the conference and possibly even be a guest speaker... I'm sure that the attendees would like to hear from developers and contributors about their experiences the past year with 1.4 as well as what's happening with 1.6 beta. -MC ___ -- 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 Asterisk ready for Prime-Time?
John You have raised few valid points. Thanks. However, I will say that it is not asterisk but people/company deploying it. Generally speaking after deployment, and as long users are using the system normally, no reboot is required. And yes, running the whole thing from standard PC based desktop will eventually cause issues hence an solid state appliance is a way to go :) Agreed. The simple fact of the matter is that most key systems and hybrids that hang on the wall and just work are mostly or completely solid state. I've been in the PBX/Key/Hybrid business since 1994 and my experience is, I'm sure, similar to most phone system veterans: keep your solid state stuff clean and cool and it pretty much never breaks; the stuff that breaks almost always seems to involve moving parts and/or the power supply. (Power = heat = eventual breakage.) Like John, I've pulled out systems that have worked for 10-15 years and never broke, they just got old. One other salient point is that the operating system and resident hardware are factors that must be taken into consideration when running a computer-based phone system. Great software running on a great OS running on crappy hardware will lead to problems. Crappy software running on a great OS running on rock solid hardware will lead to problems. (You get the idea.) To get back to the OP's question about Asterisk being ready for prime-time: it all depends. Your experience with the small systems working great but the larger one having issues isn't uncommon. I would suggest asking around on list to find out what kind of hardware is being used by those who've had lots of success, especially if you're connecting to the PSTN, because that adds yet another layer of complexity. BTW, if you're asking for my opinion, I'll give it: no, I personally don't think Asterisk is ready for prime-time in a mission-critical application. I don't use it for anything mission-critical. (For those who feel I've just blasphemed, please direct my opinion to /dev/null.) -MC That is my experience. Regards, Senad ___ -- 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] Zaptel for 1.6-beta1
Is there a minimum zaptel and libpri version for use with 1.6-beta1? Thanks, MC ___ -- 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] Difference between Asterisk and FreeSwitch
what is the difference between FreeSwitch and Asterisk , The main difference in functionality is that FreeSwitch is a voip-switch only. Technically, FreeSWITCH is a soft-switch, or a modular media switching library that can switch more than just voice. Also, technically, FS is a library, and there is a freeswitch application built on that library. The best analogy I can think of is the application curl, which is a command line app built on libcurl. Asterisk is a full-featured PBX that can do many of the things a true soft-switch can do. FreeSWITCH is (or will be, depending on your viewpoint) a full-featured soft-switch that can do many of the things that a PBX can do. It does not have any method to interface to the PSTN, other than through using another host which does have that connectivity, such as an Asterisk- based host. To be fair, this isn't quite accurate. FreeSWITCH can interface to PSTN via PRI or analog FXS/FXO using Digium, Sangoma, PIKA, etc. cards. (Any Zaptel-compatible cards should work. I've done PRI with a Tor2 clone.) Also, to be fair, the PSTN interface, like the rest of FS, is still young and therefore subject to the usual (and unusual) bugs that inhabit beta releases. Technically, the FreeSWITCH project is at RC1. The PSTN mod to FS is called OpenZAP and it is probably better described as beta. (Not an official statement, just my personal observation formed from my personal usage. I've got an Asterisk box sitting right next to a FS box and I've been playing with both of them and I can tell you that right now Asterisk is much more ready for PSTN usage.) whitch one is more scalable and reliable? That is going to depend completely on what environment you're deploying it, what features you're using, etc. Keep in mind that Asterisk is going into its third major release cycle, while FreeSwitch is still undergoing public betas and has not yet had a single general release yet. Also, note that the installbase, developer base, and userbase are all much larger, by an exponential factor, for Asterisk than for FreeSwitch, and Asterisk has a company backing it which is willing to provide commercial support. FreeSwitch, as best as I can tell, has no such support structure. These are all true. The bottom line is that FreeSWITCH is a young, but very cool, project headed by a small core development team. The lead developer is a huge Asterisk contributor - Anthony Minessale. (Check the karma page and I think you'll find he's way near the top...) The community is also small but growing quickly. There are a lot of people who use both FS and * because they have different target applications and different strengths and weaknesses. If you need a tried-and-true app that is well-supported and documented then Asterisk is an easy choice. If you are comfortable on the cutting edge or if you like the way FS is built or the way it approaches the handling of certain challenges then FS is something you should check out. This is one area where FOSS is so cool - you can totally check out both projects and give them a test drive without paying a penny in software costs. To the OP I recommend that you investigate both projects and see if one fits your needs better, which I believe is Tilghman's advice as well. -MC -- Tilghman ___ -- 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] Annoying PRI Channels Restarting Message
Is there a reason it resets? Aka does it serve any kind of purpose? Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are you using? Also, which carrier? Finally, have you turned on PRI debugging to see if it is the telco that is requesting the restart? In some cases the telco will send out a PRI message like 'service' (i.e. service request) to which the CPE will need to respond with a service ack message. Not all telcos behave the same with respect to so-called maintenance messages, so you might want to follow up with the carrier just to be sure nothing is wrong. Probably nothing is wrong but it can't hurt to check. -MC P.S. - the messages might be annoying, but if you've ever had PRI issues then those messages become comforting! ___ --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] Interface with NEC NEAX 2400
Is there anyone out there who has tried to connect up an asterisk box to make and take calls through a NEC NEAX 2400 using Q.sig or anything like it? Can anyone tell me if it is possible? Phil, I've successfully connected my NEAX 2400 to Asterisk using line side and trunk side T1's. I've only documented the line side setup: http://voip-info.org/wiki/index.php?page=Asterisk+NEAX2400+LineSide I've never tried using a PRI card though... HtH, MC ___ --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] Free T1 Card?
Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC ___ --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] Free T1 Card?
http://www.pikatechnologies.com/ -- Kristian Kielhofner Thanks, I guess I wasn't hallucinating! ___ --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] libdundi?
I would have thought an LGPL version wouldn't be out of the question. I hope not! LGPL is perfect for library-ish FOSS. Releasing libraries under standard GPL, while making Richard Stallman's heart go pitter-patter, limits what they can do since they can only go into other GPL projects. The LGPL is a great license that balances software freedom/protection with the flexibility to be used in all sorts of software projects, including (gasp!) commercial and (double gasp!) proprietary ones. A libdundi that could be included in other OSS telephony projects would definitely be a good thing. -MC ___ --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] CDR
If anybody thinks they have a magic spell that will calm down the CDR's, I will not mind the information at all! Murf, I don't know if it's relevant or not, but I do know that at least one legacy PBX vendor (NEC) has a 'solution' that helps with some of the sillier CDR's that could get generated. They have what they call a pseudo-answer timer which is basically just a way of saying, If a call doesn't last for at least X number of seconds then it really isn't a call and no CDR should be generated. It is a bit of a case of throwing away all really short phone calls, even legit ones, but it does also get rid of the silly stuff: I pick up, get dial tone, then hang up or I pick up, dial ext 1234, let it ring for two seconds and then hang up. What I'm wondering is if there's a way to apply this kind of logic to some of the scenarios you're dealing with. The hard part, I'm assuming, is having a way to make it customizable for people who have certain needs, i.e. they rely on the borked behavior, not to mention the butterfly effect (fix one thing over here, then something seemingly unrelated over there breaks). Question: is there a minimum CDR duration, be it a hard-coded value or a setting somewhere? I'm just curious if there's a way to tell the system, Look, it's safe to ignore any 'phone calls' that are less than 2/3/4/5 seconds in duration, just drop the CDR in those cases. Of course, we don't want the system to be too 'helpful' for its own good. If my minimum call duration is 4 seconds but my system can detect a busy in only 2 seconds, I don't want that CDR dropped just because the system is good at detecting congestion... I sympathize with your dilemma! I hope kicking these ideas around will at least help grease the wheels for getting a viable solution in place. -MC ___ --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] extensions.conf vs. AEL
I just got the 2nd edition Asterisk book from O'Reilly, and was surprised to find nothing in there about AEL, except a mention of extensions.ael on page 471. This is too bad. A preliminary chapter, an intro into AEL, why it's valuable, etc. would have been very welcome. Even an appendix of a few pages with examples and references to on-line documentation would have been helpful. I don't think I want to wait for the 3rd edition. Perhaps the Asterisk Cookbook will have some AEL stuff in it... -MC ___ --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] extensions.conf vs. AEL
You know, you don't have to wait for the 3rd edition... you could always write something yourself and post it on the web, or join the (mostly dormant, unfortunately) Asterisk Documentation Project. :-) -Jared Well, I could, if I _could_! :) I was hoping to learn AEL from the new book... my expression was meant as a lament for the * community. The TFOT book(s) is very cool and AEL would have made it even better. I respect the TFOT authors and a chapter or section from them would have been most welcome. I'm certain they would have done a much better job than I ever could have! -MC ___ --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] Which Asterisk version to use?
I'm a complete newbie to Asterisk and have been reading through documentation and sites for the last couple of hours trying to understand what to do to start my learning curve with Asterisk, and am very confused. It's a big world, so take a deep breath and don't worry about being overwhelmed at first. For starters, what is the difference btwn the 1.2 and 1.4 branches of Asterisk? I can't seem to find a document that describes the changes. Secondly, what is the best way to start off with Asterisk? Should I install a Linux distro from scratch and then install Asterisk on top of that, start with AsteriskNOW and go from there, or start with Tribox? What advantage do I get installing Linux/Asterisk vs. installing AsteriskNow or Tribox and starting my learning curve from there? It would seem as the most reasonable to start with a prepackaged appliance installation - no? One advantage to using the prepackage method is that you get straight into what Asterisk can do while skipping most of the how-do-I-set-it-up drama. I personally like the Trixbox distro for getting a quick setup into operation. It is relatively easy to get started with and let's you play with the system. Once you get your feet wet then it's a good exercise to learn the steps of doing a manual install. Can someone please explain the difference between AsteriskNow and Tribox? They seem to be filling the same need - a one-step easy installation of Asterisk on a brand new PC. Am I missing something? Both have GUIs, but TriBox seems to be more complete with more features. Is this not correct? Trixbox is, essentially, Asterisk + Asterisk 3rd party add-ons + decent preconfiguration. I'm not sure about AsteriskNOW. Just remember that Asterisk has an ecosystem, so there are lots of different things you can plug into it and lots of different apps that can interface with it. Just take it slow and steady and you'll do very well. Thanks so much for any information to help set me on the right path. As you can see, I am extermely confused and lost in the maze of Asterisk docs and struggling to find a little headway here. Someone already mentioned it, but get the O'Reilly book - Asterisk - The Future Of Telephony. (Abbreviated TFOT in many places.) Be sure to get the 2nd edition if you're going to buy it off the shelf. It should be out any time if it isn't already available. -MC ___ 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] Digium acquires Switchvox
I also think this is a positive thing for the Asterisk community as well, as key pieces of the Switchvox system will be rolled into the open-source version of Asterisk. (I've personally heard of two or three things that the Switchvox team has done to improve Asterisk, and I'm sure there are lots more I'm not aware of yet.) Thanks for the update. Many of us are curious as to what those features are and when they might be made available. We are looking forward to hearing more... -MC ___ 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] asterisk as a softswitch
www.freeswitch.org http://www.freeswitch.org/ (still in early beta) _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Quitoriano Sent: Friday, August 24, 2007 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch What is a good softswitch that is also open source rather than asterisk? On 8/24/07, James Jones [EMAIL PROTECTED] wrote: Yes you could, but asterisk was designed to be a PBX. I would not use it as soft switch due its limitations. It really depends on how much traffic you are going to be passing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 24 August 2007 1:11 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch Mark Quitoriano wrote: Can i use asterisk as a softswitch? This thread has been discussed over and over. Search the archives, there are more thoughts and opinions there than you probably have time or desire to read. Thanks, Steve Totaro ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. ___ --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 -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/register http://www.spreadfirefox.com/?q=user/registerr=19441 r=19441 ___ --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] FW: The trixbox Revolution Continues! Sign up forthe Webinar.
Not sure what all the licensing in TrixBox is but if they dump the open, can't we always just fork. I have not played with TrixBox in some time but most of it was just a bunch of separate, valuable projects meshed together. I don't really see how they can close that. Yes! That's one of the redeeming qualities of the GPL. They *can't* close GPL code and still redistribute it. Also, you are correct in the assessment that TB is essentially a bunch of separate, valuable projects meshed together. The value of TB, in my personal experience, is that it was way easy to get Asterisk going with lots of bells and whistles. I still use TB; I simply clean up the dial plan and a few other files. TB does a fantastic job of 'getting it all to work properly' right out of the box. If you need all of the bells and whistles, and if you don't want to get your hands dirty learning the ins and outs of all of the add-ons, then TB is a fantastic Asterisk package for you. In the end, if TB/Fonality tries to close things up I'm sure that there will be enthusiastic community members who will pick up the FOSS torch and carry on with a similar project. -MC ___ --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] FW: The trixbox Revolution Continues! Sign up forthe Webinar.
Perhaps. I'm interested in knowing what this is all about. Hopefully it's just Fonality trying to create a new revenue stream, kinda like Digium did w/ ABE. I'd hate to see them dump the open part of their community. That community is very valuable for beta testing, giving feedback, establishing an installed base of TB system, educating lots of TB-knowledgeable technicians... Let's hope for the best. -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, August 08, 2007 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FW: The trixbox Revolution Continues! Sign up forthe Webinar. Hmm beginning of the end of free trixbox by the sounds of it. It was good while it lasted but time to download the latest iso while it's still available by the sounds of it. Regards, Dean Collins Cognation Pty Ltd mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). _ From: trixbox [mailto:[EMAIL PROTECTED] Sent: Wednesday, 8 August 2007 2:00 PM To: Dean Collins Subject: The trixbox Revolution Continues! Sign up for the Webinar. http://echo4.bluehornet.com/ct/ct.php?t=1839203c=1686662876m=mtype=1 h=24C5C6D927F8E6A426872B034CFC94F4 trixbox The Official Unveiling You've been hearing about it for a while. And now it's finally here. Introducing the next evolution of the trixbox product family: trixbox Pro. What exactly is trixbox Pro, you ask? We'll tell you...next Monday. Sign up for the Webinar where we'll show you trixbox Pro in action and all the possibilities it brings you. We can only accommodate the first 1000 trixboxers on the webinar so sign up today and call in early! August 13 @ 9:00 AM PDT (16:00 UTC/GMT) http://echo4.bluehornet.com/ct/ct.php?t=1839204c=1686662876m=mtype=1 h=24C5C6D927F8E6A426872B034CFC94F4 Fonality, Inc. Fonality | 200 Corporate Pointe Suite 350 | Los Angeles, CA 90230 | www.trixbox.org http://www.trixbox.org/ This message was intended for: [EMAIL PROTECTED] You were added to the system May 3, 2007. Click http://echo4.bluehornet.com/subscribe/source.htm?c=bhKnjO3S0caW.email= [EMAIL PROTECTED]cid=105ab23bdcfebb496b833bf4db11024f here for more information | Unsubscribe http://echo4.bluehornet.com/phase2/survey1/survey.htm?CID=ypniytaction =update[EMAIL PROTECTED]_mh=8316fb751136f25ce5ff1f7461654597 http://echo4.bluehornet.com/imagelibrary/N-1686662876-979C4976A0579E088 A711D4CD3EC3723.jpg ___ --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] PRI Reset
This freaked me out at first also, but it is totally normal, and is a good way to let the telco know that your equipment is 'still alive and kicking'... -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Wednesday, August 08, 2007 7:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Reset Absolutely normal, yes. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) - Original Message - From: Jeremy Mann mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Wednesday, August 08, 2007 10:29 AM Subject: [asterisk-users] PRI Reset Is it normal for a PRI to reset the inactive B channels periodically(like once every hour). I'm seeing on my asterisk console successful restarts, just curious as this is all new to me. _ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by http://www.mailscanner.info/ MailScanner, and is believed to be clean. _ ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . Fair indeed. I would guess that a completely stable 1.2 w/ security maintenance is acceptable to the majority of users. Those folks still using 1.0.x certainly aren't clamoring for new features! The great many folks using 1.2 are happy w/ a stable release and don't necessarily need new features. A lot of those folks might consider moving to 1.4 when the stability issues and bugs are worked out. Possibly there are features that they would like to have but they don't want to invest the time and effort into a migration until they are reasonably confident that 1.4 will meet their needs. I think that having the development team be able to focus the majority of their attention on improving 1.4 is better than having them split their time between the old and new releases. I'm feeling like there's more ROI to be had improving 1.4. -MC ___ --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] Asterisk as a call recorder for ISDN30 ?
Mike, First, what kind of T1 card(s) do you have? (Just curious.) I've seen two different theories of operation, although I have experience only with one, and that's not with Asterisk. One is a passive tap, the other is a pass-through. I can't say that I know if they work or how, but hopefully someone one the list can give you a hint. A passive tap looks something like this: PSTN ---+--- PBX --- LAN | | V Asterisk --- LAN We have this exact scenario with our call recording system, except that we have a proprietary solution. (Very expensive, very yucky, I wish I had Asterisk available when we were looking for this solution... :( ) I know this works, at least in theory, because that's what we're doing right now. What I don't know is whether or not the various T1 cards that work with Asterisk can be configured for this scenario. Essentially the call recorder device (Asterisk, in the above scenario) is set up for inbound only communications. I don't know what is done at the hardware or device-driver level to configure the cards to work only on the receive pair and not the transmit pair. Also, the LAN connection is just to illustrate the fact that you may need to have some sort of on-line database to index the calls, e.g. the call on channel 3 of span 2 at 11:19am was made by extension number 123... In other words, it is useful to have a correlation between the CDR from Asterisk and the CDR from the PBX. (This is a project in its own right.) I've heard also of the pass-through scenario, which I imagine would look like this: PSTN --- Asterisk --- PBX This scenario would require two Asterisk T1 ports for each span. However, I'm guess that Asterisk and the dialplan could handle the pass-through and call recording relatively easily given enough CPU muscle and the fact that there wouldn't need to be any transcoding. At the very least the dialplan would have to handle inbound calls from the PSTN and route them to the PBX accordingly, and it would, of course, have to have some means of handling outbound calls from the PBX to the PSTN. I'll bet there are dialplan gurus out there who've done this sort of thing already... If you have then we'd love to hear about it. Please let us know what hardware you used, how you did the dialplan, and what challenges you had to overcome. Thanks, MC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Tuesday, May 29, 2007 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk as a call recorder for ISDN30 ? Mike, This thread might be of aid: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180994.ht ml -- Alex On Tue, 29 May 2007, Mike Dent wrote: Hi, would it be possible to use Asterisk to record calls only? There would be an existing PBX and calls come in on a ISDN30 line? The Asterisk box would need to sit between the incoming ISDN 30 circuit and the existing PBX. Is this possible? thanks Mike ___ --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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct + +1-678-954-0671 ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a former consultant for these customers. Agreed. Another reason not to keep futzing with 1.2. It ain't broke (except for .18 in your case), so don't keep fixing it, other than to patch security issues. (Security patches are not supposed to cause core dumps... I hope that the difference for you between .15 and .18 isn't a big security threat.) Ideally there is a recent release of 1.2 that is truly stable for the majority of users. If not, then I dearly hope that Digium get one out there before August 2007. Abandoning 1.2 while it is in an unstable condition would be a tragedy. -MC ___ --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 detection
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gommidh Riadh Sent: Wednesday, May 23, 2007 3:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax detection Gommidh Riadh wrote: For exemple I call number 0123456789 - if it is a fax then redirect to extension A - if it is a line then redirect to exention B whats ia want its somthing like AMD application that i use for the answering machine . http://www.voip-info.org/wiki/view/NVFaxDetect I've been curious about this as well. It's almost as if we need an version of app_amd that can test for all three conditions: 1 - human 2 - answering machine 3 - fax machine Has anyone heard of such a beast, or have you otherwise found a good workaround when needing to detect all three of the above conditions? -MC ___ --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] OT: USB T1/E1 Interface?
To everybody: Thanks for your thoughts and suggestions. This will be my last post to this list on this subject. I've started a blog about my research into this project: http://myossjourneys.blogspot.com/ If you want to discuss this any further please do so over there. Thanks again! -MC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, May 03, 2007 1:39 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: USB T1/E1 Interface? Way cool product. Way too cool for my neighborhood -- the interface box is $7k. Software will set you back $3k to $30k. And then I would have no clue what to do with it. Maybe we could interest the guy thats building his own open telco hardware: http://www.rowetel.com/ucasterisk/pr1.html He seems to have the skills :) On Thu, 3 May 2007, Jorge Mendoza wrote: http://www.gl.com/laptopt1.html Jorge Michael Collins wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Just for fun. I'm a telecom geek and having a USB T1 interface would be a fun toy to tinker with. Besides, it might lead to some useful products. -MC ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] OT: USB T1/E1 Interface?
Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Just for fun. I'm a telecom geek and having a USB T1 interface would be a fun toy to tinker with. Besides, it might lead to some useful products. -MC ___ --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] OT: USB T1/E1 Interface?
I can well understand the idea of having USB T1 adapters since that way you can colocate 1U Asterisk systems ;-) which at least doubles you density in a rack... Frank I'm glad I asked the question! I was just thinking to myself that it would be cool to have a USB T1 adapter so that I could tinker, but you guys have already come up with several real-world applications! I think I will research this some more and let you all know if anything interesting pops up. -MC ___ --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] OT: USB T1/E1 Interface?
http://www.gl.com/laptopt1.html That's the first item I found when I did a Google search. It prompted me to ask the question - is there something more generic than this? I was quoted a price of US$8000 for this, which is way more than I'm willing to pay for an item which would be used for tinkering, learning, FOSS-type fun, etc. -MC ___ --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] OT: USB T1/E1 Interface?
Maybe we could interest the guy thats building his own open telco hardware: http://www.rowetel.com/ucasterisk/pr1.html He seems to have the skills :) I'm working on it right now! :) -MC ___ --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] OT: USB T1/E1 Interface?
How about PCMCIA and 2 T1/E1/J1 interfaces? http://www.utelsystems.com/instruments/hardware/pist-2mp-pro.php Nice, but less portable than a USB - most desktops and servers don't have a PCMCIA slot. I'm thinking about the 'U' in USB. If I'm going to have something be portable, why not make it work with as many different systems as possible? -MC ___ --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
[asterisk-users] OT: USB T1/E1 Interface?
Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I've seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Thanks! -MC ___ --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] Nagios asterisk monitoring
On Wednesday 11 April 2007 03:11:38 pm Brandon Kruse wrote: I wrote a very extensive plugin for cacti to monitor asterisk. It uses the manager interface to poll and get statistics for 1.4 and 1.2. Let me know if you interested, ill post it, or email me directly. -bkruse I did not appreciate how cool this was until I researched RRDTool and Cacti! I am definitely interested in this as well. I have a feeling that many in the * community will want to learn more about this. Thanks, MC ___ --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] Verizon-Vonage Lawsuit
Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- You ain't kidding!!! Next thing you know someone will try to patent this: User picks up communications unit human interface device, a.k.a. 'handset', in response to audible ringing indication (visual 'ring' indication is optional). Just when I thought I couldn't have a lower expectation for a government agency - here comes the USPTO. Monumental foolishness. -MC P.S. - in broader terms, are there any of these patents that threaten FOSS telephony projects? ___ --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] Sendmail and exchange for voicemail integration
Jordan, I don't know if you've down this step before, but my network admin sent me these instructions a few months ago. It allows you to tell your Exchange Server's SMTP to allow relays from specific domains, hosts, or subnets. Hope it helps. (Works for Exch 2000 and 2003.) -MC 1. Go to Exchange System Manager 2. Drill down to Servers, (your Exchange server), Protocols, SMTP, Default SMTP Virtual server. 3. On Default Virtual Server, right click on it, select properties. Select the Access tab on the top, then select the Relay button. 4. On the Relay Restrictions window, make sure the Only the list below button is selected. 5. Add an allowed IP, subnet or domain name 6. When done, hit OK 3 times and that's it. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Friday, March 23, 2007 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sendmail and exchange for voicemail integration Jordan, Assuming that the voicemail users are email users on the domain for exchange then your DNS entries for MX will take care of most of the work. Sendmail on the Centos installs I have done has required no changes to the default config to work with our exchange servers. You probably will want to make sure that the SMTP protocol on Exchange allows the Sendmail server to relay. On 3/23/07, Jordan Novak [EMAIL PROTECTED] wrote: I am having real trouble getting Asterisk to send to exchange. They are on the same LAN. Does anyone know of a walkthrough for this setup. I have gotten it to work before, but that was to a hotmail account. Jordan Novak ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --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] Problem with ATT Maintenance protocol inPRI connection, no B+D channels available
span=1,0,0,esf,b8zs,crc4 This needs to be span=1,1,0,esf,b8zs I'm not sure if the crc4 is necessary. Doug I concur with Doug. I have two PRI's in one system. My zaptel.conf looks like this: span=1,1,0,esf,b8zs # PRI line - LD Qwest (interstate) bchan=1-23 dchan=24 span=2,2,0,esf,b8zs # PRI line - SBCLD (intrastate/local) bchan=25-47 dchan=48 HTH, MC ___ --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] Asterisk Automated Outbound Messaging
Looks like user interface is not a concern - if they are thinking of FTP text files. In this case, a simple script to kick off some call files should suffice. Won't take a week. (Search for call file.) But having to deal with answering machines is always tricky for any automation. Yuan Liu Don't forget the other 'fun' issues related to auto-dialing with .call files (or AMI originate): Detecting and handling fax machines Figuring out whether a 'failed' call is a no answer or an invalid phone number (Yes, this is a tricky one, especially when using PRI) Getting correct CDR info back into the host system, if this is a requirement Establishing the calls is the easy part. Figuring out exactly what transpires AFTER the calls are originated is the true challenge. -MC ___ --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] Problem with ATT Maintenance protocol in PRIconnection, no B+D channels available
I've never seen a PRI dchannel on a T1 on a timeslot other than the 24th. Are you sure that it's really on channel 23? I think he meant channel 23 of channels 0~23, aka the 24th channel. -MC Matthew Fredrickson On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote: Thanks for your answer, Bruno. However, the configuration you provided is for an E1 connection and we are using a T1, having channel 23 as D channel. Bruno De Luca [EMAIL PROTECTED] escreveu:d-channel is in midle bchan=1-15,17-31 dchan=16 loadzone = it defaultzone = it Kanelbullar wrote:Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? * Could there be any configuration issue on our side? * Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0)
[asterisk-users] Replacement Wiki - options (Formerly 'status of voip-info')
I would suggest that we create a new wiki, make it solely for Asterisk topics, as not to offend or replace voip-info. Build mirrors to multiple sites and multiple domain names. This would give this community a second resource with redundancy which is what I think ALL of us are looking for. I have taken the pleasure, of registering the domain name ASTERISKONLINE.ORG. I would like to know what the community feels about an Asterisk-only wiki. I can see pros and cons of Asterisk-only vs. Asterisk/FreeSwitch/Yate/OpenPBX/etc. My gut says keep it open for everything OSS/VoIP. (I have no logical reason for feeling that way - it's just a gut feeling.) I will donate a dedicated server with bandwidth to the cause. I am looking for additional people to help populate the wiki with useful information and to help maintain the site. I would suggest that ee have maybe 4 or 5 mirrors to start off and a core group of admins to help maintain the site. Thanks for putting your money where your mouth is! This is the kind of action the community needs. I am willing to work with anyone else that is about providing a solution to our current issue. If you guys want to REALLY work toward a solution, here's the chance. For the individuals that are interested in helping e-mail me. I hope you get some respondents. In the meantime it might be good to check out the fledgling wiki here: http://www.voip-wiki.us It uses MediaWiki which has a nice, clean interface and seems pretty easy to use. -MC ___ --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] Asterisk Auto-dial out
I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 # I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Is there a way that I can just put in the number and have the system decide the channel to use for calling it? What I would like to do: Channel: #=== This number could be # 7645 in which case go via SIP/7645 # 68001 which should go to CiscoSIP/68001 # 12127778866 which would go via Zap/G2/12127778866 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Based on dialing plan the system should be able to route the call to whatever channel supports dialing that number. You probably want to use the Local channel. Definitely hit the wiki and check it out: http://www.voip-info.org/wiki/view/Asterisk+local+channels The idea behind the local channel is that you can, in effect, drop a call right into a specific part of the dialplan. From there, your dialplan can handle the logic of figuring out which technology and channel to use. -MC ___ --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] Rx+,Rx-,Tx+,Tx- of TE110P
Hi everybody, i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P and also if you can tell me have to made a cable like that?? Modem Teleco ---Self CrosscableAsterisk You might check this out for a quick reference: http://www.voip-info.org/wiki/view/crossover+T1+cable -MC ___ --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] Digium TE110P
Hello, i've installed trixbox with TE110P TDM400B, but no led is ON in the TE110P, i don't know why even if the 4 leds of My TDM are greens any explaination Thank You No LEDs on TE110P (and similar cards) can sometimes mean that the Zaptel driver isn't running. Can you run zttool and see anything happening, even red or blue alarms? Also, have you been able to confirm that your drivers are even loaded? Do: lsmod and make sure you have your drivers: zaptel, wcte11xp I don't personally have a TE110P so I can't offer you any advice specific to this card... Let us know what happens. -MC ___ --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] PRI Call Start
Yeah, it's hard to know what it would be filed under. However, if you use zap trunks then you'll want to know about this page: http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+Channels BTW, see Dialing a Group for specifics on 'g' vs. 'G' as well as other cool stuff. -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, February 13, 2007 1:45 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Call Start Oh interesting. I don't recall seeing that documented anywhere. Thanks! On 2/13/07, John Novack mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: g hunts low to high G hunts high to low John Novack Matt wrote: Hi, If I have a PRI with 23 channels on it.Can I setup Asterisk to start outbound calls at 23 and hunt back to 1? I know I can individually do it with gX/23/5551212 (or something along those lines). But is there a way to make it hunt FROM 23 down to 1. By default it starts at 1 and hunts up to 23. ___ --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 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] PRI Call Start
At times I think the wiki has grown out of control. I hear you. I'd pay money to anyone willing to create and maintain a master index! -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, February 13, 2007 7:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Call Start Thanks good info on that page.At times I think the wiki has grown out of control. There is almost too much info there... that even with a search engine you can miss some. Oh well.. Thanks for the pointer. On 2/13/07, Michael Collins [EMAIL PROTECTED] wrote: Yeah, it's hard to know what it would be filed under. However, if you use zap trunks then you'll want to know about this page: http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+Channels BTW, see Dialing a Group for specifics on 'g' vs. 'G' as well as other cool stuff. -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, February 13, 2007 1:45 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Call Start Oh interesting. I don't recall seeing that documented anywhere. Thanks! On 2/13/07, John Novack mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: g hunts low to high G hunts high to low John Novack Matt wrote: Hi, If I have a PRI with 23 channels on it.Can I setup Asterisk to start outbound calls at 23 and hunt back to 1? I know I can individually do it with gX/23/5551212 (or something along those lines). But is there a way to make it hunt FROM 23 down to 1. By default it starts at 1 and hunts up to 23. ___ --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 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 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] Trixbox vs. Custom install
Of course, you should take this with a grain of salt since I tried [EMAIL PROTECTED] (now TrixBox) for a total of 2 weeks before gutting it. Now, I just use my own GUI for everything from graphical setup to scripting. There is nothing wrong with starting out with Trixbox. I still use it because I like the Linux distro (CentOS) and I like the fact that it sets up lots of stuff that I don't have to bother with. I used Trixbox to learn a lot about how to use Asterisk, then I went back and did a clean install on a separate machine to learn about setting up and installing Asterisk. For me, having a working system first, playing with it, breaking it, etc. was very useful because it gave me perspective when setting up a system from scratch. Now I actually have two systems to play with: one Trixbox and one scratch * install. (I get the best of both worlds, but I have nothing in production just yet. I'll decide later which way to go once I'm doing playing with my two 'sandboxes.') Bottom line is this: you need to start somewhere. Would you rather start by using a working system or by building from the ground up? Neither way is perfect for everyone. If you have the luxury of doing both then I can highly recommend it - each method has taught me valuable lessons that the other method didn't. HTH... -MC ___ --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] Using Local Channels with Originate
(Sorry for top-posting)
I'm making good progress. However, so as not to clutter the list I will
post my solution on the wiki in the next few days. I'll send out the
link as soon as I've got something substantial for you to review.
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian K.
Alexander,Jr. (Vision Point Systems)
Sent: Tuesday, February 06, 2007 6:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Using Local Channels with Originate
Ack... That should be I am using analog for the proof of concept but
plan to use PRI for the actual system...
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian K.
Alexander,Jr. (Vision Point Systems)
Sent: Tuesday, February 06, 2007 8:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Using Local Channels with Originate
Right now I am using analog but the plan is to use PRI for the proof of
concept but the actual system would use PRI. I know that the analog
support is supposed to be somewhat unreliable but I have yet to get it
to detect even a busy - not even once. I can only assume that I missed
some setting somewhere but I can't find it.
I am curious to learn more about your solution. If you post more
information I might be able to help you with your RD. In any event
thanks for posting up and in advance for keeping us posted on your
progress.
-Brian
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Collins
Sent: Monday, February 05, 2007 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Using Local Channels with Originate
I haven't quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out...
The trick to using the DIALSTATUS seems to be to put it in the handler
for the h (hang-up extension).
[outdialer]
exten = 100, 1, Dial(${numberToDial})
exten = h, 1, Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,NoOp(Answered)
exten = s-BUSY,1,NoOp(Busy)
exten = s-NOANSWER,1,NoOp(Not answered)
exten = s-CANCEL,1,NoOp(Cancelled)
exten = s-CONGESTION,1,NoOp(Fast busy)
exten = s-CHANUNAVAIL,1,NoOp(Channel unavailable)
[dialerplan]
exten = s,1,Background(demo-congrats)
exten = s,n,WaitExten
so on ...
Here are the manager commands I am using:
Action: login
Username: test
Secret: nottelling
Action: originate
Channel: Local/[EMAIL PROTECTED]/n
Context: dialerplan
Extension: s
Priority: 1
Variable: numberToDial=ZAP/4/1234567890
Action: logoff
I am always getting ANSWERED for ${DIALSTAUS} so something is not quite
right. Hopefully I am getting closer.
Brian,
What kind of Zap hardware/telco lines are you using? I am using PRI and
I am able to get a dial status in the hangup extension. The problem I
run into is that I get NO ANSWER as the hangup cause even for invalid
phone numbers... I also get cluttered CDR's. In the meantime I'm
working on a solution that I hope will give the best of both worlds.
I'm relying on the API events instead of local channels. I'll post more
information when I've made more progress. However, I've made 2500 test
calls and I haven't lost a single 'OriginateSuccess' or
'OriginateFailure' event. (I'm keying on these, specifically the
'OriginateFailure' event because it has a 'Reason' value that gets
populated: 0=Invalid, 3=No Ans, 5=Busy.)
Hope to have more info posted this week.
-MC
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RE: [asterisk-users] Using Local Channels with Originate
I haven't quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out...
The trick to using the DIALSTATUS seems to be to put it in the handler
for the h (hang-up extension).
[outdialer]
exten = 100, 1, Dial(${numberToDial})
exten = h, 1, Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,NoOp(Answered)
exten = s-BUSY,1,NoOp(Busy)
exten = s-NOANSWER,1,NoOp(Not answered)
exten = s-CANCEL,1,NoOp(Cancelled)
exten = s-CONGESTION,1,NoOp(Fast busy)
exten = s-CHANUNAVAIL,1,NoOp(Channel unavailable)
[dialerplan]
exten = s,1,Background(demo-congrats)
exten = s,n,WaitExten
so on ...
Here are the manager commands I am using:
Action: login
Username: test
Secret: nottelling
Action: originate
Channel: Local/[EMAIL PROTECTED]/n
Context: dialerplan
Extension: s
Priority: 1
Variable: numberToDial=ZAP/4/1234567890
Action: logoff
I am always getting ANSWERED for ${DIALSTAUS} so something is not quite
right. Hopefully I am getting closer.
Brian,
What kind of Zap hardware/telco lines are you using? I am using PRI and
I am able to get a dial status in the hangup extension. The problem I
run into is that I get NO ANSWER as the hangup cause even for invalid
phone numbers... I also get cluttered CDR's. In the meantime I'm
working on a solution that I hope will give the best of both worlds.
I'm relying on the API events instead of local channels. I'll post more
information when I've made more progress. However, I've made 2500 test
calls and I haven't lost a single 'OriginateSuccess' or
'OriginateFailure' event. (I'm keying on these, specifically the
'OriginateFailure' event because it has a 'Reason' value that gets
populated: 0=Invalid, 3=No Ans, 5=Busy.)
Hope to have more info posted this week.
-MC
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[asterisk-users] RE: [SOLVED] Dial option G - Passing parameters?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Thursday, February 01, 2007 12:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial option G - Passing parameters?
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers (Customer)
Customer identifies himself, and now I use Dial w/ the G option:
Dial(Zap/g9/${agentext}|60|mG(Agent_Xfer^s^1)
Customer hears MOH while the Dial app gets the agent on the line
My destination context looks like this:
[Agent_Xfer]
exten = s,1(Customer),Meetme({$ConfRoom}|qM)
exten = s,2(Agent),Macro(Connect_to_agent,${Customerid},${ConfRoom})
Customerid and ConfRoom are channel variables that are set in the
Originate action and at the start of the dialplan processing,
respectively.
The idea is to put the customer in a conference room, listening to
MOH,
until I can get an agent on the line. (This part works pretty well.)
The agent is an extension on a legacy PBX, so a simple Dial with a
macro
has undesired side effects. (Specifically, the customer hears ringing
or the legacy PBX's MOH, depending upon the status of the transfer.)
Putting the customer in a conf room, listening to music, is the best
solution I can think of.
The problem is that I don't know how to get the two channel variables
over to the Agent leg of the call. I don't see anything in the docs
about the G option accepting arguments to pass to the called leg. Is
there any way that I can get the two variables' values over to the
called leg?
-MC
FYI,
After some researching I realized that I did not understand variable
inheritance. I've 'globalized' the two variables in question so that
they are inherited by the second leg of the call. For your reference,
the helpful information was found on the wiki:
http://www.voip-info.org/wiki-Asterisk+variables
(Specifically under the heading Inheritance of Channels Variables.
-MC
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RE: [asterisk-users] API Originate Action - distinguishingbetweenNoAnswer and Invalid phone number
I have been having a very similar problem. Has anyone here gotten a DIALSTATUS for calls started with originate? I did some research and saw some posts that local channels are the solution to this problem. However, I could not find examples of how to use local channels with originate. I could not get it to work. I posted a topic (Using Local Channels with originate) to this list yesterday with the details about what I had tried. Maybe you will see what I missed. -Brian Brian, I have had zero success with local channels as well. When I dial a local channel, I actually get TWO outbound channels. It's weird. My logs show two passes through the dialplan even though I've called Dial(Local/xxx) only once. The phone number received two calls simultaneous. I've tried with and without the /n just to see if there's a difference. (There isn't, at least on my system.) If anyone out there has success stories using local channels with API Originate (or .call files) then we'd love to hear about it! Please let us know how you've overcome the limitations of autodialing, i.e. no DIALSTATUS, no dialplan processing on failed attempts unless you have a 'failed' extension, no DIALSTATUS information going to the 'failed' extension, etc. I still don't know how to distinguish between a legit NO ANSWER and an INVALID phone number. (They both 'fail' on an Originate and they both produce a second CDR with a disposition of NO ANSWER. BTW, I'm using PRI, and I've tried both inband and outofband signaling.) Thanks, MC ___ --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
[asterisk-users] Dial option G - Passing parameters?
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers (Customer)
Customer identifies himself, and now I use Dial w/ the G option:
Dial(Zap/g9/${agentext}|60|mG(Agent_Xfer^s^1)
Customer hears MOH while the Dial app gets the agent on the line
My destination context looks like this:
[Agent_Xfer]
exten = s,1(Customer),Meetme({$ConfRoom}|qM)
exten = s,2(Agent),Macro(Connect_to_agent,${Customerid},${ConfRoom})
Customerid and ConfRoom are channel variables that are set in the
Originate action and at the start of the dialplan processing,
respectively.
The idea is to put the customer in a conference room, listening to MOH,
until I can get an agent on the line. (This part works pretty well.)
The agent is an extension on a legacy PBX, so a simple Dial with a macro
has undesired side effects. (Specifically, the customer hears ringing
or the legacy PBX's MOH, depending upon the status of the transfer.)
Putting the customer in a conf room, listening to music, is the best
solution I can think of.
The problem is that I don't know how to get the two channel variables
over to the Agent leg of the call. I don't see anything in the docs
about the G option accepting arguments to pass to the called leg. Is
there any way that I can get the two variables' values over to the
called leg?
-MC
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[asterisk-users] API Originate Action - distinguishing between No Answer and Invalid phone number
I've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no answer and an invalid? For me, a 'failed' attempt is dialing an invalid number, and I'd like the CDRs to reflect that. I'd like a no answer to be as 'successful' as a busy. I'm familiar with the 'OriginateFailure' event and it's 'Reason' field, but I don't know how to get that reason into the CDR. Is that possible? Thanks, MC ___ --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] API Originate Action - distinguishing between NoAnswer and Invalid phone number
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork
Sent: Thursday, February 01, 2007 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] API Originate Action - distinguishing between
NoAnswer and Invalid phone number
On 2/1/07, Michael Collins [EMAIL PROTECTED] wrote:
Is there a way to distinguish between a no answer and an invalid? For
me, a 'failed' attempt is dialing an invalid number, and I'd like the
CDRs to reflect that. I'd like a no answer to be as 'successful' as a
busy.
The ${DIALSTATUS} channel variable stores the result of the dial attempt:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
You can store it on the CDR's userfield column using the cdr function:
Set(CDR(userfield)=${DIALSTATUS})
Actually, I can't. The dialplan execution goes straight to the 'failed'
extension. When it does so, the DIALSTATUS variable gets cleared out. I have
this in my dialplan:
exten = failed,n,Noop(Dial status is '${DIALSTATUS}')
The log yields this:
-- Executing NoOp(OutgoingSpoolFailed, Dial status is ) in new stack
Is there perhaps a way to make DIALSTATUS persist or get populated when the
dialplan hits the failed extension?
-MC
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RE: [asterisk-users] Getting confused on signalling mode Vs framingand encoding: T1 CAS
ESF B8ZF Inbound = EM Immediate Outbound sig =Wink Start Yield to Glare = Yes In zaptel.conf, when having something like span=5,0,0,cas,b8zs and in zapata-channels something like signalling=featb try em_w: E M Wink Start Jerry is right - you need to set signaling in zaptel.conf like this... signalling=em_w ... so that it matches what's in zapata.conf. 'featb' is a reference to 'Feature Group B' which is for ISDN and not for good ole CAS/RBS. -MC ___ --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] vxml support
Can Asterisk support vxml? Can i work with Asterisk and vxml? Is there any AGI framework that can use vxml? It seems like support is still a bit limited, but evidently it is available: http://www.voip-info.org/wiki/view/VoiceXML HtH, MC ___ --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] setting up AMD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't
see how to get AMD to print out more. I have it call and say Hello like
I normally would. I've tried to say more and less doesn't seem to
matter. After I hangup it does recognize hangup. Here's logging during
an attempt where I make outbound call and answer, but then hangup after
1-2 seconds:
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
[8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
[5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
The amd.conf:
[amd]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
Peter,
It looks like your initial silence setting might be having trouble. The
amd.conf file has a value of 3500 but the log file is showing 8000. Try
changing the amd.conf to something like 3000 and issue a reload at the
CLI. Make another test call and see if the trace still shows 8000 for
the initial silence. I think having an initial silence value that is
longer than the total analysis time might be causing the undesired
behavior.
Let us know what happens when you try to modify the initial silence
value.
-MC
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RE: [asterisk-users] setting up AMD
Hmm... not too sure what's up with this one. I've only used AMD with
Zap channels, so I don't know if there are any hidden gotchas with using
SIP.
Has anyone else used app_amd with SIP calls?
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up AMD
now I have amd.conf set to this:
initial_silence = 3700
greeting = 2500
after_greeting_silence = 1200
total_analysis_time = 6000
min_word_length = 100
between_words_silence = 50
maximum_number_of_words = 4
silence_threshold = 860
The resulting log is this:
Jan 24 18:53:04 DEBUG[31555] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]
Jan 24 18:53:04 VERBOSE[31567] logger.c: -- AMD:
SIP/sip.broadvoice.com-087743a0 6079362172 (null) (Fmt: 4)
Jan 24 18:53:04 VERBOSE[31567] logger.c: -- AMD: initialSilence
[3700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime
[6000] minimumWordLength [100] betweenWordsSilence [50]
maximumNumberOfWords [4] silenceThreshold [860]
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Scheduled a registration
timeout for sip.broadvoice.com id #19
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 104:
Match Found
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Registration successful
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Cancelling timeout 19
Jan 24 18:53:28 DEBUG[31555] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED] '
Jan 24 18:53:32 DEBUG[31555] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 24 18:53:36 VERBOSE[31567] logger.c: -- AMD: HANGUP
Jan 24 18:53:36 DEBUG[31567] app_amd.c: Got hangup
Jan 24 18:53:36 DEBUG[31567] pbx.c: Extension s, priority 2 returned
normally even though call was hung up
Jan 24 18:53:36 DEBUG[31567] chan_sip.c: update_call_counter(6079362172)
- decrement call limit counter
On 1/24/07, Michael Collins mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't
see how to get AMD to print out more. I have it call and say Hello like
I normally would. I've tried to say more and less doesn't seem to
matter. After I hangup it does recognize hangup. Here's logging during
an attempt where I make outbound call and answer, but then hangup after
1-2 seconds:
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
[8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
[5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] '
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
The amd.conf:
[amd]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
Peter,
It looks like your initial silence setting might be having trouble. The
amd.conf file has a value of 3500 but the log file is showing 8000. Try
changing the amd.conf to something like 3000 and issue a reload at the
CLI. Make another test call and see if the trace still shows 8000 for
the initial silence. I think having an initial silence value that is
longer than the total analysis time might be causing the undesired
behavior.
Let us know what happens when you try to modify the initial silence
value.
-MC
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RE: [asterisk-users] Detecting Disconnected Numbers - PRI
The correct way to determine the ending cause of a call is the
${HANGUPCAUSE} variable that Dial creats. Just to be sure, set
priindication=outofband in /etc/asterisk/zapata.conf. HANGUPCAUSE
should always be set.
HANGUPCAUSE is indeed always set. The question is, Set with what data?
The problem is that the telco doesn't consistently and uniformly send
back the Q.931 hangup cause. Believe me, I've pored over mountains of
Q.931 logs, both with inband and outofband signaling. The telcos just
plain suck at delivering this information consistently. They usually
get it right, but when you are making tens of thousands of dial attempts
per day and the telco is giving you accurate info 90% of the time then
you still have 100's of call records with suspect data. Garbage in,
garbage out.
My work around is to make multiple attempts on so-called invalid numbers
and to keep track of the results. If I dial a phone number and get
hangup cause 16 less than two seconds after the dial attempt, and if I
can repeat that result, then I assume it is truly a disconnected or
otherwise invalid number.
-MC
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RE: [asterisk-users] Detecting Disconnected Numbers - PRI
original message I am trying to automatically detect disconnected numbers when using the outbound dialer I have written. * Some numbers hang up immediately with a Cause Code 0 and no voice treatment * Some numbers get voice treatment with a PROGRESS indication and an associated Cause Code 0 * Some numbers get voice treatment with a PROGRESS indication and no associated cause code (CC=0) My application can pick up the PROGRESS indication (if I get one) and handle the hangup, but not if I don't get a cause code! Is there anything I can do to ensure that I always get a PROGRESS indication with cause code or a hangup with cause code? Behaviour of the PRI seems to differ across telcos and also across numbers. I don't want to just assume hangup on PROGRESS indication as this may not be a disconnected number - it might be a forwarded or redirected number. I need to achieve consistency and this is proving very difficult. Has anyone else had this issue and if so, which tree should I be barking up? /original message Yep, I experienced this frequently. I have several PRI vendors and they all give me the same line of crap: Well, PRI is good, but it's not perfect... Sad but true. I feel comfortable in saying that there is no 100% guaranteed way of detecting disconnected numbers on a PRI. I've done lots of testing and come to the conclusion that you have to do your best to work around it. For example, I know that phone number xxx-yyy- is disconnected. I dial it 25 times with Asterisk. 18 times I get one cause code (like 'invalid' or fast busy - I don't recall the exact cause code number), 6 times I get PROGRESS indicating ring-no answer and 1 time I get traditional busy. All calls to same phone number, same provider, made one right after the other. Oddly enough, if I call the number on a POTS line I *ALWAYS* get the disconnect message. It's one case where advanced technology yields poorer results than the old stuff. I know that doesn't help but I wanted you to know that you're not alone. -MC ___ --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] answer machine detection
One thing that is really confusing me at this point: if I want to
leave
an automated answer machine message, and amd tells me it's a machine,
how do I know when to start leaving the message ? Some intros are
long
(thanks for calling, me and mine are not here right now, please leave
a
message after the beep) and some are short Leave a message.
Is there a way of waiting in the dialplan for a beep or something like
that ?
Excellent question. I've been experimenting with the 'WaitForSilence'
app. I've not tried to detect a beep since answering machines and
voicemail systems will not be uniform in their beep sounds.
I've used this with reasonably good success:
[lmtc]
; if detect ans machine, come here and leave a msg to call back
exten = s,1,Answer
exten = s,n,Wait(5)
exten = s,n,WaitForSilence(1000,2)
exten = s,n,Playback(Not-right-party-live-Eng)
exten = s,n,Wait(.3)
exten = s,n,SayDigits(${dnum}) ; Supplied by Originate action
exten = s,n,Wait(1)
exten = s,n,Playback(Not-right-party-live-Eng)
exten = s,n,Wait(.3)
exten = s,n,SayDigits(${dnum}) ; Supplied by Originate action
exten = s,n,Wait(1)
exten = s,n,AppendCDRUserField(${cdrdelim}Y)
exten = s,n,Hangup
As soon as I detect AMD, I goto lmtc,s,1. I wait 5 seconds, then do
wait for silence. I've experimented with various settings, and I
settled on wait for two occurrences of 1000ms of silence. This is a
reasonable balance between having a two second pause at the very
beginning of the message that I leave and accidentally starting my
message playback too early because of silence detected during the target
machine's outbound message. Sometimes you have a message like, High
this is so-and-so. pause Please leave me a message. That pause can
sometimes trip up your WaitForSilence app if you don't wait long enough
for silence.
In my case, I'm leaving a message that says, Please call us at phone
number and provide reference number dnum. I repeat the message just
in case I started playing it too soon the first time. Thus far I've had
pretty good success. YMMV, so tinker with the WaitForSilence settings.
If you're okay with a two second pause at the beginning of the message
that you leave on the target answering machine then these settings will
probably work for you.
Let us know how it goes.
-MC
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[asterisk-users] .call files no longer generating CDR files
I've got a curious one: all of a sudden my .call files and my manager API 'Originate' actions are no longer producing a CSV file. The call still generates just fine, and Master.csv is updated. However, I don't get the usual CSV file in the form of xx.csv where xx=account number. I didn't make any changes that I'm aware of. Is there something to check? I'm on 1.2.12, and this machine was working fine just a few days ago... Any insights would be much appreciated. -MC ___ --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
[asterisk-users] RE: [SOLVED] .call files no longer generating CDR files
I've got a curious one: all of a sudden my .call files and my manager API 'Originate' actions are no longer producing a CSV file. The call still generates just fine, and Master.csv is updated. However, I don't get the usual CSV file in the form of xx.csv where xx=account number. I didn't make any changes that I'm aware of. Is there something to check? I'm on 1.2.12, and this machine was working fine just a few days ago... Any insights would be much appreciated. -MC On a hunch, I rebuilt Asterisk and Asterisk-addons from the source and everything started working again! Big sigh of relief... -MC ___ --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] HowTO configure voice T1
David is correct: there are several issues to resolve. Some common T1 settings in the USA are: Framing: ESF Line coding: B8ZS These are very common settings. Now you'll need to find out what the signaling type is. No point trying to guess - they vary greatly. If you can find out how the previous piece of equipment was configured the you'll have a great starting point. -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion Sent: Thursday, January 04, 2007 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HowTO configure voice T1 T1s can use many different signalling types. You need to find out which one is running, what the line encoding is, etc. PRI vs T1 are not the only distinctions... On 1/4/07, Mark Greene [EMAIL PROTECTED] wrote: Alright guys here is my question. What is do I need to set switchtype, and signalling to in zapata for a voice T1. This is not a PRI. I cannot say that enough. It is NOT, A, PRI. It is just a Voice T1 with 24 voice channels. There is not a D Channel. It runs from one office to another and USED to plug into two opt. 11c but now one end is going to plug into an asterisk box. - Mark ___ --Bandwidth and Colocation provided by Easynews.com http://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
[asterisk-users] (OT) Where to post free source for AGI?
Also, anyone have suggestion on licensing? LGPL? FreeBSD? One advantage of LGPL over GPL is that GPL is 'viral' whereas LGPL is not. For a more in depth discussion please see: http://www.ugcs.caltech.edu/manuals/devtool/autotoolset-0.11.4/toolsmanual_87.html In short, if you want anyone to be able to distribute your software within their own packages, even proprietary and/or commercial ones, then use the LGPL. If you want your software to follow the tenets of 'free and open source' more strictly, then use the GPL. Both licenses protect your software, but they place different limits on how the software is distributed. Hope this helps! -MC ___ --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] Best Hardware for Asterisk Server?
I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. One word of caution: some have had various hardware issues getting certain telephony cards to work with certain Dell PowerEdge servers. If you aren't going to have telephony cards in your system, i.e. VoIP-only setup, then you're probably good to go. If not, do a list search on Dell PowerEdge and review the feedback given by those who've already been where you are now. Hopefully their experience will save you time, money, and the occasional headache! -MC ___ --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
[asterisk-users] Double quotes in CDRUserField?
Question: I'm trying to put a double quote into the CDRUserField. What I end up with is a pair of double quotes. Example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField() exten = s,n,AppendCDRUserField(moredata) My record will look like this: datamoredata What I want is: datamoredata The wiki mentions using a backslash in order to 'quote the character' as it says. However, this example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField(\) exten = s,n,AppendCDRUserField(moredata) Yields the same results: datamoredata Is there something that I'm missing? Thanks, MC P.S. I'm using CSV for my CDR's ___ --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] Double quotes in CDRUserField?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Tuesday, January 02, 2007 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Double quotes in CDRUserField? Michael Collins wrote: Question: I'm trying to put a double quote into the CDRUserField. What I end up with is a pair of double quotes. Example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField() exten = s,n,AppendCDRUserField(moredata) My record will look like this: datamoredata It's common for CSV files to escape quotes by putting two of them to indicate it is a quote within the string, not the end of the string. Perhaps you could accomplish what you're going for with something else, say an underscore character? Regards, Trevor Peirce Under the circumstances I think that is the easiest thing to do. I can do some minor shell scripting to handle the parsing of the userfield. Thanks for the suggestion. -MC ___ --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] Dialed Number missing from the CDR when using callfiles.
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Lyman Sent: Saturday, December 30, 2006 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialed Number missing from the CDR when usingcallfiles. *snipped Second, when using a .call file (or the manager interface's Originate action) the 'Dial' action is executed BEFORE entry into the dialplan, so if it fails, nothing in your dialplan is executed and you get a somewhat *snipped not *exactly* true. you need to add ;this extension MUST be here for OriginateFailure triggers exten = failed,1,Hangup to your context used for *send too after connect* The one caveat here is that * actually cuts two CDR's for the call. This isn't normally a problem unless half the data you want is in CDR one and half is in the other! :) I have done some scripting to extract the relevant data from each record and condense it back down to one - a small price to pay to have the functionality that I really need. -MC ___ --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] Dialed Number missing from the CDR when usingcallfiles.
you need to add ;this extension MUST be here for OriginateFailure triggers exten = failed,1,Hangup to your context used for *send too after connect* Richard, THANK YOU!! This makes a lot of sense - I don't know why I didn't catch that before. I can add my SetCDRUserField stuff in the 'failed' extension. -MC ___ --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] Dialed Number missing from the CDR when usingcall files.
I think the CDR generator of the Asterisk
needs change to record the complete information.
Agreed. However, there are still challenges here. First, you could use
the custom_csv to create your own CDR layout that includes the dialed
number, but you'd still need to come up with a way to get that into
MySQL. (As far as I know, there isn't a custom_mysql module.)
Second, when using a .call file (or the manager interface's Originate
action) the 'Dial' action is executed BEFORE entry into the dialplan, so
if it fails, nothing in your dialplan is executed and you get a somewhat
complete CDR, except there's nothing in the CDRUserField.
If you aren't worried about calls failing then you can use the
CDRUserField to store the dialed number. I'm using Trixbox and the
MySQL stuff is already configured to store the uniqueid and the CDR
userfield.
The call file will need something like this:
SetVar: dialednum=5551212
Then you'll need a dialplan entry like so:
exten = s,n,SetCDRUserField(${dialednum})
This is a workaround that a lot of people use because they don't need
the CDR userfield for anything special. Personally, I put tons of stuff
in the userfield and just delimit my items, like this:
exten = s,n,SetCDRUserField(${dialednum})
exten = s,n,AppendCDRUserField(:${firstname})
exten = s,n,AppendCDRUserField(:${lastname})
exten = s,n,AppendCDRUserField(:${misc1})
exten = s,n,AppendCDRUserField(:${misc2})
Then I'll have a field at the end of the CDR that looks like this:
5551212:firstname:lastname:misc1:misc2
Technically I use the cdr-csv for my CDR's, but Trixbox turns on MySQL
automatically so I get MySQL CDR's also. (I just manually clean them
out every month or so.)
HtH helps! Let us know if you make any progress.
-MC
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RE: [asterisk-users] Dialed Number missing from the CDR when using callfiles.
The CDR, both the csv file and in MySQL does not contain the dialed number (src) in case of a call placed using .call files. Is this is Bug ? The cdr should have complete info, what ever the source or method of the call. I have found this same problem and have not found a solution within Asterisk. AFAIK, the CDR subsystem simply does not put the 'dialed number' in the record. Not a 'bug' so much as an unfortunate design choice. Another issue is that when an auto dial call (i.e. at .call file or manager interface 'originate' action) fails, the CDR record is cut BEFORE any dialplan entries are executed, so you can't put this information into the CDR UserField via the dialplan. The wiki implies that you can use the local channel to bypass this limitation. I've tried it, but I cannot get it to work. (I always end up with two channels bridged together when all I want is one channel dialing out to deliver a message to the called party.) The wiki stuff is here: http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels If anyone has figured out how to use the local channel to initiate an autodial out call, please respond. I'd love to see how it works. -MC ___ --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] Asterisk 1.4 - no PRI and no Zap?
But apart from that: have you tried at least building that driver with 1.4.0 ? Yep. The build process seems to work just fine. The ztcfg and zttool stuff all acts normal. I copied tor2.c and tor2-hw.h from the custom 1.4.0-beta1 drivers (that work just fine with asterisk 1.4.0-beta1) and recompiled zaptel, libpri, asterisk and asterisk-addons in that order. My concern is that the custom drivers might have one or more lines changed in zaptel.c or something else. I tried a diff but there was way too much there so I bailed. I've asked the OEM to let me know when the 'official' 1.4 drivers are ready. Thanks, MC ___ --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
[asterisk-users] Asterisk 1.4 - no PRI and no Zap?
Has anyone else installed the official 1.4.0 release? I have, and it installed very easily. However, I don't have any of my usual command line tools for monitoring and debugging zap channels and PRI lines: asterisk1*CLI pri show span 1 No such command 'pri show' (type 'help' for help) asterisk1*CLI Ditto with zap stuff: asterisk1*CLI zap show channels No such command 'zap show' (type 'help' for help) asterisk1*CLI I didn't see these in the 'core' commands either. Any thoughts? Thanks, MC ___ --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] Asterisk 1.4 - no PRI and no Zap?
You must use zaptel 1.4 and libpri 1.4. Asterisk 1.4 specifically has checks in the configure script to check for the unique stuff in those versions and the associated channel driver (chan_zap) will not build without it. I think I found the issue. My Tor2 clone has a modified driver. The latest driver they've produced is for Zaptel 1.4.0 beta1. I'll wait until they produce an official Zap driver version 1.4.0. Thanks, MC ___ --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] Answering Machine Detect (AMD) time values
Does anyone know what the time values in amd.conf are? Are they seconds, fractions of seconds, heartbeats, what? Milliseconds. ;'initialSilence' is the maximum silence duration before the greeting initial_silence = 25; Maximum silence duration before the greeting. It doesn't say in amd.conf or at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] Re: Match a Numer - then continue with, dialplan
Firstly, in the setup you are envisaging, how do you distinguish which company the caller is calling from? Their extensions number? The context at which they enter the dialplan? Or something else? Good questions, all of them. Unfortnately, I don't have answers to them. I wanted to take our 3000 line python script, which we'd used due to inadequacies of the dialplan, and throw the horrible nasty thing out the window. Secondly, how do you distinguish between destination numbers in one company from those in another? Number range? Context? Tony, Thank you for asking the appropriate questions! I think you've gotten to the crux of the matter. Doug, take some Advil and read the rest of this post tomorrow! :) At my work, we have a saying that we use when trying to figure out how to overcome some technical challenge. It helps us to focus on the solution, not the problem. We simply ask, What is Utopia? Then, in plain English, we describe the perfect world. (Choose the language of your locale for this exercise.) Doug, could we try this exercise? Could you collate the bits and pieces of your posts in this thread and distill them into a point-by-point description of your Utopia? Use as few technical terms and Asterisk-specific references as possible. A good starting point is the list of to-do's that you do for each call: Do they have voicemail? Do they have feature ABC? Do they have feature XYZ? After listing all of that, then give us the description of what needs to happen next, the part about deciding which caller ID info to send. Pretend like you're explaining it to a bunch of idiots who understand only small words and short sentences. :) My gut tells me that the solution lies in the big picture, not the details. The more eyes that see the big picture the better. Thanks for your patience! -MC ___ --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] zapata.conf zaptel.conf
I am configuring two cards in Trixbox. 1 TE110P T-1 card and one TDM2400P with 16 fxs ports (All 24 show up in zaptel.conf so the PRI channels start at 25). Can I use a channel range to separate the config for each card, as shown below, or do I have to enter configs for each channel? Also in zaptel.conf I see that the TDM card is span 1 board zero, And the T-1 card is Span 2 board 0. So for the T-1 card I entered span=2,1,0,esf,b8zs Is this correct? What are the trailing 1,0 for after the span ID? Check out: http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax OT-In reference to posting messages-what is top posting? Top posting means putting your reply at the very top of your post and leaving the thread contents below. You'll notice that I left your original post mostly in tact, cutting out only the boring email header info. It is proper etiquette not to top post but instead put your replies at the very end of the post so that those reading it can see it in chronological order. If everyone top posted then the quoted thread would be in reverse chronological order and you'd need to scroll to the bottom to see the start of the discussion and then scroll up as you read. Most of us prefer to scroll down as we read! :) -- ; Zapata telephony interface [trunkgroups] ; [channels] ; context=incoming switchtype=national ; signalling=fxs_ks usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default channel=1-16 ; context=incoming switchtype=national ; signalling=pri_cpe usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default channel=25-47 ___ --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
[asterisk-users] Fax machine detect (akin to AMD)
Has anyone done any fax machine detection on outbound calls? I've heard of NV's fax detect app but I haven't seen any indications that it supports outbound fax machine detection. -MC ___ --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
[asterisk-users] AMI - Originate Action and Busy, NoAnswer calls - CDR
Gang, I'm wondering if anyone has run into this problem and found a solution. When I use the manager interface to generate a call, I don't get very much information in my CDR records when the dial status is BUSY, FAILED, NOANSWER, etc. I am putting the dialed number into the CDR Userfield in my dialplan, but the field doesn't populate the CDR record unless the Originate action is successful and the dialed party answers the call. I need to postprocess the CDR records and I absolutely have to have the phone number in the CDR. Ideally I'd like to populate the CDR Userfield with several pieces of information, which I am able to do only if the Dial() or Originate operation results in a connect. I've tried numerous variations of context/extension wrangling to no avail. I can supply examples of what didn't work but I'm really interested in hearing about examples that do work. Has anyone found a workaround or a best practice that allows CDR records to contain the dialed phone number for every Dial() or Originate that Asterisk processes? Thanks, MC ___ --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] Auto dialing: .call file vs. manager interface
The manager interface expects Exten NOT Extension argument header. Well honk my hooter! I had been using 'Extension' but since I always used the 's' extension I never noticed anything goofy until I tried a numeric extension. Thanks for the heads up. -MC ___ --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
[asterisk-users] Auto dialing: .call file vs. manager interface
Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good). The error I get in the log is as follows: Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 failed so falling back to exten 's' Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 still failed so falling back to context 'default' The autodial_start context looks like this: [autodial_start] exten = _9.,1,Noop(Looks good) exten = _9.,n,Goto(dialout,s,1) The dialout context just has the call handling stuff, AMD, etc. It works when the Goto works, but the Goto only seems to work when using a .call file and not the manager interface. The .call file looks like this: Channel: Zap/g0/5596221408 Callerid: 5597337550 MaxRetries: 0 RetryTime: 30 WaitTime: 30 Context: autodial_start Extension: 95596221408 Priority: 1 Account: 5898832 Has anyone experienced this issue and/or found a way around it? Thanks, MC ___ --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
[asterisk-users] Digium TE407P vs. Sangoma A104d
Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is 5 stars! Awesome! It Rocks! They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I'm interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --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] asterisk manager originate command
I want to know how to get the uniqueid or a call started from asterisk manager using Originate command. Are you wanting the uniqueid for the call right after it is started, i.e., while it is still in progress? What is in your Dial command? -MC ___ --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] asterisk manager originate command
There is no dial command, I'm sending originate action from asterisk manager. Oops, I didn't ask my question correctly. You're right, it isn't a dial command. What I wanted to know was the contents of your originate action, e.g.: Channel= 'zap/g0/' . $dialed_num (From one of my Perl scripts using POE::Component::Client::Asterisk::Manager) Thanks, MC ___ --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] Connecting Asterisk to an NEC Aspire
New Asterisk user, wondering if anybody has connected an Asterisk box to an NEC Aspire S? We're in the beginning processes of attempting this, we'd like to have the Asterisk box connected as an extension off of the NEC box, wondering about the wiring and settings/programming needed to get the units talking to each other. Analog or digital stations? -MC ___ --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] Connecting Asterisk to an NEC Aspire
On the NEC, digital stations (ip1na-12txh) I am not familiar with the Aspire, but if it is even remotely like the 2400 then you might be able to get a jumpstart using my 2400 how-to: http://www.voip-info.org/wiki/index.php?page=Asterisk+NEAX2400 It deals with getting a Tormenta2 clone talking to a 2400 station side T1 card. If you know an NEC tech familiar with both PBX's then he might be able to translate this into something you can use. Hope this helps and good luck! -MC ___ --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] Asterisk 1.4 : App_Swift (Cepstral) Howto
Great link. After I all you said I get this error loading the module in asterisk via load app_swift The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. [Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error loading module 'app_swift': libswift.so.4: cannot open shared object file: No such file or directory [Nov 30 13:54:08] WARNING[7825]: loader.c:607 load_resource: Module 'app_swift' could not be loaded. Any ideas? The README file reminds you to do this: Install one of the Cepstral Voices. Use the standard install directory /opt. On Linux don't forget to insert /opt/swift/lib into your /etc/ld.so.conf file and run ldconfig. Make sure you've got /opt/swift/lib in your ld.so.conf file! -MC ___ --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] What's up with the Manager Interface?!?!
Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Component::Client::Asterisk::Manager. These have abstracted away the lowest level of programming. I know you've done Python in the past - I hear that there's a module for AMI called py-Asterisk. Have you seen or tried that? Ditto with Ruby - a module called RAMI. Both are on sourceforge. Also, could you hum a few bars about what you're trying to accomplish with your API? I'm curious about the big picture. Thanks! -MC ___ --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
