Re: [Asterisk-Users] [PRI] TE110P
On Jun 14, 2005, at 9:07 PM, Michael L. Young wrote:Does the TE110P support NI1 or NI2? (I think the answer is both)What is the number of digits outpulsed? Is there a version number on the TE110P card? Switchtypes supported on asterisk (from zapata.conf.sample):; national: National ISDN 2 (default); dms100: Nortel DMS100; 4ess: ATT 4ESS; 5ess: Lucent 5ESS; euroisdn: EuroISDN; ni1: Old National ISDN 1; qsig: Q.SIGNumber of digits outpulsed is configured as part of your dialplan, i.e. how many do they want... and if its a PRI:; unknown: Unknown; private: Private ISDN; local: Local ISDN; national: National ISDN; international: International ISDNThere is the type of Dialplan usually Unknown and then their end should "figure it out" like it does on a normal POTS line.As for version number I don't know, but that is probably less important then they think because so much of the cards "functions" are done in software in Asterisk itself or the Zaptel driversLater;Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID - B8 Message
On May 27, 2005, at 6:50 AM, Andrew Kohlsmith wrote: On May 27, 2005 03:20 am, Nathaniel Angelo A. Torres (247talk) wrote: Any idea how I can generate a B* message on the asterisk box (out of order) message? Easy. Do *not* have an exten = line in your PRI incoming context that matches the number you want SIT to be played for. At least for Bell Canada PRIs, * will return something along the lines of no number here and Bell will take care of notifying the calling party. e.g. if your DID block is 555-1000 to 555-1010 and you want 555-1001 to appear out of service: exten = 5551000,1,DoSomething() exten = 5551002,1,DoSomething() exten = 5551003,1,DoSomething() exten = 5551004,1,DoSomething() exten = 5551005,1,DoSomething() exten = 5551006,1,DoSomething() exten = 5551007,1,DoSomething() exten = 5551008,1,DoSomething() exten = 5551009,1,DoSomething() exten = 5551010,1,DoSomething() (note the total absence of 5551001) Or you can have additional control (If your using the CVS version) with: exten = 5551001,1,SetVar(PRI_CAUSE=1) exten = 5551001,2,Hangup For more info see: http://www.voip-info.org/wiki-Asterisk+variable+PRI_CAUSE Later; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Hi; Just to a a little perspective. The closest equivalent Dialogic board (i.e. connects 4 T1's to a pci bus) is just under $7000 USD per board. Now while this isn't comparing apples to apples since the Dialogic board has more onboard processing services it does give a reference data point. The Digium board at just under $1500 USD looks quite good based on that comparison... Later; Tim On Apr 7, 2005, at 12:53 PM, Craig Guy wrote: [rant] I wish my local reseller would 'dump' the product, or at least offer it cheaper without support. The digium PRI cards IMHO are way too expensive for those of us who are familiar with them and are only interested in warranty support. I will probably soon be buying another 5 of them and spending over $A10,000 in the process. The cards are more expensive than the server they're going into (Dell poweredge 750's). When a GPL'd hardware design costs more than an entire proprietary server (including chassis, motherboard, dual hard disks and remote access card) then there is something very wrong in the market. I do not possibly see how a quarter length PCI card should cost more than an entire rack mount server. IMHO bring on the competition, Asterisk should divorce itself from Digium, the sooner the better. Asterisk is a software product and should stand alone and not be subsidised by the hardware. Marks salary should come from selling trainig and Asterisk support services, not hardware. If Digium gets money from selling Digum hardware, where then is the incentive for Asterisk to support alternative hardware (BRI for example). Imagine if Linus was employed by Intel, Linux would only be an empty shell of its current self with no support for embedded platforms, Motorola CPU's, WRT54G's, etc. [/rant] Craig - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 7:43 PM Subject: RE: [Asterisk-Users] Sangoma VS. Digium On Thu, 7 Apr 2005, Matteo Brancaleoni wrote: I hate to say that, but the problem is that Digium doesn't do this. They allow resellers to do market dumping, by not imposing fixed list prices to resellers, they also compete with they're own distributors/resellers by offering the cards online and by offering services directly to end users. In this way they're destroying they're own reseller network and there's no commercial gain into supporting the end user (as resellers). Resellers are almost universally a useless money-sink. Most add no value at all, they are simply another logistics point. Distributors, on the other hand, are usually very knowlegable and are able to support their customers (the resellers) quite well. My advice: always *always* buy from as early in the channel as possible. Prices are better and the support is _way_ better. Of course, if you are not familiar with the problem space for which you are purchasing a solution then resellers can add a lot of value. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Slightly OT - Snom 190 function keys via subscribed config
Hi; Well for the firmware go to: http://www.snom.com/download/share/ To see how it wants the settings, manually configure a phone and look what the Settings tab at the bottom of the bottom of the Lefthand side menu shows... Later; Tim On Mar 9, 2005, at 9:35 PM, Shaun Dwyer wrote: Hi All, I realise this is off topic, but its likely the best place to ask! I sent an email to snom support a few days ago but have yet to recieve a response.. Perhaps some one has found a solution to this problem already? I've searched the mailing lists and google and found nothing useful. I've also read Snom's mass deployment documentation but thats no real help in this case. Cheers, -Shaun Original Message Subject:Snom 190 function keys via subscribed config Date: Tue, 08 Mar 2005 11:15:18 +0800 From: Shaun Dwyer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Hi We have bought 5 snom 190 hardphones and I'm in the process of setting up some dynamic configuration scripts for the phones to pull their config from via HTTP. I'm having problems passing settings to the phone with the '' character in it. I want to be able to set the Function keys to specifc destinations. In the PDF file that describes how to do this, it states that the '' character is treated as a comment. Is this only the case when its the first non-white space char found on a line, or is it true for any position in the resulting data being passed to the phone. I'm currently using snom190-3.56m-SIP-j.bin firmware on the phones. A sample of the resulting output from the configuration script: === fkey0!: line fkey1!: line fkey2!: dest sip:[EMAIL PROTECTED];user=phone fkey3!: dest sip:[EMAIL PROTECTED];user=phone fkey4!: dest sip:[EMAIL PROTECTED];user=phone === The 190s seem to ignore all fkey settings. The existing setting in the phone that I have from manually configuring the phone earlier remains and dosn't become overwritten. Also, how can I get hold of the Alpha/Beta firmware for the 190s? Cheers, -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: VoIP-to-TDM processing on-card?
A couple things... Please don't post html to the list, Thanks! Second... It might help to have a little more info on your application. The quicknet cards are FXS/FXO to PCI (or CARDBus) not to VoIP. The VoIP section is done by the application, I.E. asterisk. So I believe the TE110P and TE410P/TE405P would fulfill you needs, depending on the application... Later; Tim On Jan 20, 2005, at 11:58 AM, Olson, Dana wrote: Sorry. I don't know what I'm smoking today. We need T1 interfaces... :P So let me rephrase the question: Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple T1 interfaces? The QuickNet Internet LineJack seems to meet the description I believe, but it only has a single FXS or FXO. Are there any cards that have multiple T1 ports? Thanks. __ Dana Olson HelpDesk Technician TELESPECTRUM, INC. 1-800-704-9111 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO
On Jan 11, 2005, at 1:16 PM, Wilson Pickett wrote: Thank you to all who replied, I will be trying this out as soon as I can down the system to reload the modules. There was also a fix to change the frequency in what was then wcfxs.c which is how I got our older phones to ring. Thanks again to everyone who replied, the boostringer=1 in modules.conf seems to have solved my problems! But I did have to reboot the machine, not just reload the modules as I would have thought... Later; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI RLT support
Hello; On Jan 12, 2005, at 8:17 AM, Girouard, Marc wrote: Does the PRI implementation support RLT (Release Line transfer) on a DMS 100 and DMS 250? Also can one D-channel support multiple T1? Can it also support a backup D-channel? If by RLT you mean a transfer where a call comes in on channel a, you call out to another person on channel b and connect the two and then both channels clear because the upstream switch connects the two, no. I believe there is a bounty on it though... I think its called 2 B channel transfer (http://www.voip-info.org/wiki- Asterisk+bounty+PRI+2B+channel+transfer). For your second question, rephrased... Does asterisk support NFAS with backup D channels? Yes, see http://www.voip-info.org/wiki-NFAS Later; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO
On Jan 11, 2005, at 11:42 AM, Jim Van Meggelen wrote: Peter Svensson wrote: On Tue, 11 Jan 2005, Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage does the FXS modules on a TDM400P put out? Ultimately, that depends on how much current is being drawn, but my multimeter reports about 70V on a 6 foot line cord. Generally, I'd expect an FXS to put out more like 90-110V (that's what my TalkSwitch supplies). The ring voltage can be configured. Try setting the module parameter boostringer when the wctdm module is insmod / modprobed. Which is accomplished in 1.0.x by the following: # modprobe wcfxs boostringer=1 And in CVS HEAD (I assume, as I haven't tested this) by: # modprobe wctdm boostringer=1 Who, exactly, Boo Stringer is has not yet been determined. Boo Radley? NO, Boo Stringer. (Sorry for the pun, folks, but I will now never forget this parameter name). Or adding the line: options wctdm boostringer=1 to the file /etc/modules.conf Thank you to all who replied, I will be trying this out as soon as I can down the system to reload the modules. Later; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO
Hi; I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage does the FXS modules on a TDM400P put out? Thanks; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium T100P T1 Card
On Jan 5, 2005, at 11:23 AM, Wiley Siler wrote: snip To further explain my siutation, I should give you some more background on my setup. My current setup has an AdTran 616 on the wall breaking out my 6 analog lines and delivering my data to the office. I have two TDM400P cards receiving 6 analog lines which are used for both fax and voice. I have had numerous problems with this ISP and I just want to get away as soon as possible. Problem is, I have a contract that won't expire for a while so I need to use these lines for something. The ISP wants a contract extension and some setious cash to do the upgrade. Better to just seek alternate service. I originally bought my T100P thinking I would get digital lines and all the goodies involved. Then budget constraints and an ISP that wants too much to convert me to Digital lead to a temporary solution. I would use the analog lines for a while longer. Well, that has run its course and I have to get to something more stable. The PRI card looks pretty good at this point. So getting back to the T1 PRI issue (and I am playing catch up here), my goal is to just deliver new service into this office over my T100P and just dump nothing but fax out those old lines. That way I can reserve the digitals for our truly important calls and still reap the benefit of having those old analog lines. Large Snip So to summarize: Currently you have a T1 from an ISP, this ISP is currently delivering 6 analog FXS phone ports and delivering fractional T1 internet access over the Ethernet ports on the Adtran 616. To help clear up an issue that may have confused others, all the lines you have are delivered in digital form the Adtran converts the 6 phone channels to analog. In theory (from reading mailing list not from personal exp.) the Adtran could be replaced by a Linux box with a T100P and Asterisk (probably without any config changes on the ISP end). Later; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New T100P Pri install suggestions?
1) Most PRI T-1's in the US are set up as NI2 even if the actual switch is something else. 2) Correct 3) As far as I know zttool is it in asterisk. But then w/my install it 'Just Worked' (tm) So I didn't even need that... 4) When an incoming call is presented on the PRI you will get some number of digits put into EXTEN and the dialplan will try top match it... i.e. if the DID range the telco has assigned you is (555) 555-1100 thru (555) 555-1199 they will probably just send you the last 2 or 4 digits, assuming 4 digits, you will get (and have to match in your dialplan) 1100 thru 1199. Hope that helps! Tim On Nov 30, 2004, at 9:48 AM, Lyle Giese wrote: 1) national would be my choice. 2) the number will depend on the load order of the modules when you modprobe them. I don't know the answers to the other two questions as I have not done a PRI with * yet. Lyle - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Monday, November 29, 2004 7:15 AM Subject: [Asterisk-Users] New T100P Pri install suggestions? In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a DMS500 that is compatible with several different switchtypes including 5ess, dms100, etc. * will clock sync from this pri. This system also has a tdm04b (4-port fxo) installed and working, and is expected to remain after the t100p is implemented. Zttool can see both cards at the moment. Questions: 1. Is a switchtype=national (NI2) a reasonable choice, or are there other types that others have found to be more usable/stable with * in the US? 2. Since the current system has a working tdm04b defined as fxsks=1-4 in /etc/zaptel.conf, how will I know when implementing the t100p whether those b-channels should be 5-28 or 1-23? (eg, which card has channel 1? It almost appears that defining the span= first with bchan=1-23 and follow it with fxsks=25-28 (for the tdm04b), is about the only way to do that without creating ambiguity for the reader.) Is that a reasonable approach or assumption? 3. Other then zttool, what other methods have others found useful for diagnosing layer 1 2 type issues remotely during an initial pri install? (After ensuring layer 1/2 functionality, I'm assuming 'pri debug', etc, is most appropriate for diagnosing higher layer and dialplan issues.) 4. In very general terms, when passing incoming pri calls to a specific extensions.conf context, is using 'exten = s,1...' a reasonable way to handle inbound calls initially? (Is the 'called number' passed in EXTEN like it is with sip calls?) Any other tips/tricks for handling an initial pri installation? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New T100P Pri install suggestions?
On Nov 30, 2004, at 10:36 AM, Steven Critchfield wrote: On Tue, 2004-11-30 at 10:22 -0600, Timothy Costello wrote: 4) When an incoming call is presented on the PRI you will get some number of digits put into EXTEN and the dialplan will try top match it... i.e. if the DID range the telco has assigned you is (555) 555-1100 thru (555) 555-1199 they will probably just send you the last 2 or 4 digits, assuming 4 digits, you will get (and have to match in your dialplan) 1100 thru 1199. If you are on a PRI, there is no time penalty in call setup to send the entire phone number. In fact, it is beneficial to have the full number as it reduces conflicts later on if you ever want to add numbers. Not to mention you will make your telco provider much happier if they don't have to come up with consecutive phone numbers but rather can pull from any current gaps and possibly across exchange numbers. If you are on an EM Wink where the number is transmitted via DTMF or MF, you will find a time penalty for each digit transmitted and therefore you will want to keep the number of digits low. OF course if you are choosing EM wink, it is possible you aren't wanting to use any of the advance features of the PRI and/or are hamstrung by some other limitation outside of asterisk. Very true! But most likely you will have to specifically request the 10 digit number, by default most telcos (as evidenced by prior postings and my experiences with SBC :-) will give you 4 digits. PRI is definitely the way to go if only for the speed of call setup. Other nice features are the ability to send a PRI hangup cause, so if a given DID is not in use you can signal Out of Service just like the telco... I'm sure there are many more! Later; Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/Panasonic PRI Integration
On Nov 24, 2004, at 10:09 AM, Alfred Certain wrote: snip 1- PSTN provides the E1/PRI (EuroISDN) 2- Asterisk receives the PRI 3- Asterisk provides a second PRI to Panasonic 4- Panasonic receives the asterisk PRI snip Follow is the relevant config section of the files: ZAPTEL.CONF span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 Snip Whichever of the 2 spans is from the PSTN should be providing clocking... i.e. span=1,1,0,css,hdb3,crc4 Otherwise you may get clock slips on one or the other interface... also make sure the panasonic is getting clocking from the Asterisk server. That is clocking for the PRI's should go from the PSTN connect to the Asterisk server to the Panasonic otherwise you may well run into problems like this. Later; Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New. Testing?
On Oct 22, 2004, at 9:15 AM, [EMAIL PROTECTED] wrote: Hello all. I am new to the list and after doing some research on Asterisk this week I would like to get started testing. stuff deleted... I currently have an unused ISDN (BRI) line that I was thinking about cancelling until I learned of Asterisk. I thought about buying one of the BRI PCI cards (listed on the Digium website) to use in a test server. Although, I see that they can be rather expensive for something that I most likely will have to just throw away when we move (I assume we will have a T1 or fractional T1 in the new building.). My question is, What would you guys recommend I use to get started testing/looking at this software? I only see these two (affordable) options for testing: Buy an ISDN BRI interface board, use it and test it then throw it away. (This option wouldl give me two lines to test with, which would be nice.) Buy an analog board and use a dedicated line. (I think we only have one incoming line that I could use for testing. OR, could I use the two POTS ports from our ISDN router and in effect just use the two ISDN data lines as POTS lines? In the long run I think the ISDN BRI interface board will teach you more as you will have access to full and reliable call progress and get a much better idea of how fast connections are established etc. I used the TDM400P and the X100P for my Proof of Concept testing and then switched to a ISDN PRI via T100P for deployment and it took a while to get used to and adapt to. On an analog interface dialing out onto the PSTN asterisk assumes the call is answered (unless call progress is turned on but that is still not 100% reliable). Whereas on an ISDN connection it knows what the state of the line is (i.e. Ringing / Answered / Busy etc). Also on analog incoming call you don't get DNIS so they get dumped into the s extension in the context you define. On ISDN you will get the DNIS and have to parse it in the dialplan. All of this is valuable experience for when you get the T1-PRI for the new location. It will speed up you deployment a lot! Later; Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] searching for a nifty solution for different outgoing msn depending on the sip-user
On Oct 14, 2004, at 12:03 PM, Frank Sautter wrote:
our asterisk server is currently connected via 4 isdn trunks to our
main pbx using it as a voip gateway for homeworkers.
currently this is the dial command for outgoing calls
exten = _., 1, Dial,CAPI/141:${EXTEN}
what i like to do, is giving each sip-user a different outgoing msn
(the 141 in the example above).
the only solution i found, is to put each user into a different
context, but this leads to a very complex and error-prone
extentions.conf.
is there a niftier solution?
Well I'm not sure it's niftier but how about:
exten = _.,1,SetVar(MSN=${CALLERID:7:3})
exten = _.,2, Dial,CAPI/${MSN}:${EXTEN}
Assuming that the 3 least significant digits of their callerid (from
sip.conf) is equal to the MSN you want. If not use a database get keyed
by thier sip callerid...
Just an idea...
Later;
Tim
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Re: [Asterisk-Users] New User Questions (was: Computing horsepower needed)
On Dec 11, 2003, at 1:38 AM, Steven Critchfield wrote: lots deleted If you look over the list again, you will see that questions that get ignored tend to get the entire list flamed for not being helpful. No one here tries to run people off but you will rarely see a message where we treat any one with kid gloves. If a person requires this kind of treatment, they will have to pay a vendor to deal with their system and their personality problems. As for the documentation complaint, just because there isn't a book called asterisk for dummies yet doesn't mean there isn't documentation. The current documentation requires work on the part of the newbie to get at. The wiki is fleshing out nicely. The old handbooks still exist. There is over 2 years of mailing list traffic to document the system. Learning takes time and effort. Asking questions here short circuits the time and effort. We answer these questions to help out, but sometimes our answers are meant to eliminate dead ends. Maybe they are not as nice as you or others would like, but later when you have the knowledge to share you can decide how to dole it out. Extremely few of us have received our knowledge from source that aren't otherwise available to all. So you as well as everyone else here have a great chance of becoming a peer or better yet surpassing some of us. -- Steven Critchfield [EMAIL PROTECTED] A couple comments. One reason people on the list get testy is that this comes up every 1-2 months: http://lists.digium.com/pipermail/asterisk-users/2003-November/ 027775.html and somewhere (maybe on the wiki) should be a link to ESR's How to Ask Smart Questions: http://www.catb.org/~esr/faqs/smart-questions.html I know it's been posted to the list several times. It should be part of the FAQ to read it before asking questions... Later; Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loading dialogic drivers
I've had problems with Dialogic apps using GlobalCall with similar symptoms, I had to type export LD_PRELOAD=/usr/dialogic/lib/libgc.so before running them. Maybe Mark's answer solves that problem also... Tim Need to have chan_dialogic.so = yes in the [globals] Mark On Thu, 18 Sep 2003, pedro bulach gapski wrote: I am one of those trying to use old dialogic hardware with *. I have the following error when loading the driver: [chan_dialogic.so] = (Dialogic Global Call API Support) dlopen of libicapi.so failed: dlerror=/usr/dialogic/lib/libicapi.so: undefined symbol: gcdb_InsertLinedev WARNING[1024]: File chan_dialogic.c, Line 832 (load_module): Failed to start Global Call (GC) WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_dialogic.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_dialogic.so failed! Indeed, gcdb_InsertLinedev is not defined in libicapi. It is defined in libgc, which is linked to chan_dialogic. Anyone has seen this before? [], pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Timothy F. Costello Sr. Systems Analyst Ne te quae siveris extra ObQuote: Sounds great! If I miss, I get to be captain. -- Chakotay to Janeway, about phasering an apple off her head, Coda ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
