Re: [asterisk-users] dahdi-channels.conf for Digium TDM2400
Hi, On Wed, Dec 22, 2010 at 9:49 AM, Alex Saavedra a...@masterline-logistics.com wrote: I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line=2 WCTDM/0/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by DAHDI tools, and since it's working I didn't want to change it. Is it safe to remove them? Short Answer: NO!! Longer Answer: The settings all apply to channels, which are defined by the channel = 2 directive. If I'm remembering correctly, the channel is set at the end of the Stanza, not at the beginning. So, your blank callerid and group would apply to your next channel directive (3?). Now, I remember reading there is a way to flip the channel definition bit (channel = XX) to the top of the stanza, but can't recall. Now, if in between two channel definitions you have repetition, it might be ok to trim things up, as long as it has the right information -- the last setting is the effective one. And the bit that starts ;;; is a comment, which is actually ignored by asterisk. Hope this helps, Gerald. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP can't insert - Can someone please help
Hi Bruce, On Sat, Jul 10, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote: I have my html/php file set so that the input field only takes 3 digit 3 digit 4 digit (NPA, NXX, Block) so your purposal of: *'201,0); drop database YOUR_DATABASE'; *would fail due to big length and also I tested with inputing letters and my IF function caught it and exited. Further more, everything else (other than phone input fields) is drop down boxes with specific numbers or letters inserted in them. I should be 100% safe with those right? Another moment of trepidation should be triggered when you use the words input field as related to forms. While most people will use an ordinary web browser and whatever fields you provide, hackers aren't most people. Anyone wanting to break your site isn't going to be nice and follow the nice rules and use the forms which might have validation. Even beginner not-nicers can put together a simple form with your POST as their target and whatever field lengths and values as they want. You have to treat all input as hostile, since it all can be. It's the only way you can be safe. Thanks, Gerald -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP can't insert - Can someone please help
Hi Bruce, On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce bruceb...@gmail.com wrote: Further to my last post, I added this to santize. I also created a new mysql user with access to only findmefollow portion of the asterisk table for limited access and assigned only two simultaneous connections with only 10 changes queries per hour (as I know that no more queries will be put through probably) if ($npaa=200 $nxxa=200 $npaa!=900 $npaa!=911) Should that suffice against SQL injections? The if condition changes the string to number so it removes the chance of people adding other characters and it also sticks to format NPAN or 2XX2. There are two things -- the first is, who call this script? If it's something you control 100%, you can mitigate the risk a bit. I don't really like this tact, because if the script gets repurposed, you end up with something that could be very dangerous. The second thing is simple -- most people think small here, but you have to think big and know a bit about how PHP works. PHP strings are pretty amazing things, and one of the pesky things is that you can put all kinds of things in it. Now, if that string variable is created as a result of a form input, then that string can be anything. For a moment, think about if it $npaa = '201,0); drop database YOUR_DATABASE'; Now, that is pretty nasty, and it would muck up further SQL injections, but now you get the idea. You should always check to make sure the data you are getting is what you are expecting, and exclude what you aren't. So, are your tests sufficient? I can't remember off the top of my head if the string - integer only considers the first number, or it considers the whole string. (PHP usually errs on the side of ease of use, so I think my snippet above would still pass your test). If your expecting only numbers, I'd write a function that ensures that only numbers are parts of the input. (And not just for the 3 above variables). Really, you should never see $_POST(var) (or any PHP CGI variable) that derives directly from user input. It takes a few minutes extra, but it'll save hours of sorting later if you get hit by a SQL injection. Hope this helps, Gerald -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is WARNING: Got 200 OK on REGISTER that isn't a register?
Hi Last couple of days I received the subject WARNING message on a home-based asterisk pbx. Is someone spoofing a register method on port 5060? Or, is this warning something random (sort of like sporadic alarms on an analog port)? (This warning message is from chan_sip.c). Am running asterisk V1.4.18; (the hardware is an AMD 64 X2 and a Digium 400P with 4 ports - just a home-based pbx) - and using Ubuntu Intrepid (alternate desktop version). Thanks for any reply, Gerald Harshany g...@jerryh.us ___ -- 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] astcanary not exiting in asterisk V1.6.1
Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a play machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing through the bugs/lists, I see where a patch to trunk V1.6.0.7 was made (recently) due to astcanary not exiting when asterisk dies. Is this a different scenario than just stopping and restarting asterisk (as opposed to asterisk 'dying')? So, I'm just wondering if it is too soon to expect the patch for astcanary to have been applied in svn V1.6.1? Do I need to do something? Thanks for any reply, Gerald Harshany g...@jerryh.us ___ -- 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] astcanary not exiting in asterisk V1.6.1
- Original Message - From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 15, 2009 9:28 AM Subject: Re: [asterisk-users] astcanary not exiting in asterisk V1.6.1 On Wednesday 15 April 2009 04:25:15 Gerald Harshany wrote: Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a play machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing through the bugs/lists, I see where a patch to trunk V1.6.0.7 was made (recently) due to astcanary not exiting when asterisk dies. Is this a different scenario than just stopping and restarting asterisk (as opposed to asterisk 'dying')? So, I'm just wondering if it is too soon to expect the patch for astcanary to have been applied in svn V1.6.1? Do I need to do something? If there's another release candidate, it will show up there. Otherwise, the fix will wait until 1.6.1.1. -- Tilghman OK. Thanks for the reply - will just wait then. Gerald ___ -- 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] What is WARNING: Got 200 OK on REGISTER thatisn't a register?
I understand - Thanks for the reply. Yes, I have been registering with the sip provider Voicepulse for about 2 years, but never saw the message before. In the last 2 days or so it has popped up about 5 times each of these days, which started me wondering what the messages really meant. Gerald - Original Message - From: Martin asteriskl...@callthem.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 15, 2009 9:33 AM Subject: Re: [asterisk-users] What is WARNING: Got 200 OK on REGISTER thatisn't a register? Your box receives a 200 OK message as though it would have sent the REGISTER sip message - trying to register with a sip provider as a sip device. Asterisk doesn't recognize it because: 1) the REGISTER was not sent from Asterisk 2) the 200 OK was sent too late 3) there's some other issue like NAT or so Martin On Wed, Apr 15, 2009 at 4:30 AM, Gerald Harshany g...@jerryh.us wrote: Hi Last couple of days I received the subject WARNING message on a home-based asterisk pbx. Is someone spoofing a register method on port 5060? Or, is this warning something random (sort of like sporadic alarms on an analog port)? (This warning message is from chan_sip.c). Am running asterisk V1.4.18; (the hardware is an AMD 64 X2 and a Digium 400P with 4 ports - just a home-based pbx) - and using Ubuntu Intrepid (alternate desktop version). Thanks for any reply, Gerald Harshany g...@jerryh.us ___ -- 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 ___ -- 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] credit card processing
Hello, On Sun, Sep 28, 2008 at 1:52 AM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi Guys We have a service that can be use by our customer via a website and also via telephone. [...] Do you know any company that do this ?? I recently completed implementing such an application - integrated with www.chasepaymentech.com. Contact me off-list if you are interested. Gerald. ___ -- 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] CDR accuracy
I think that the most appropriate answer for this would be it depends on your setup and requirements. Some of our customers bill all answered calls for the entire minimum duration/increment (even if duration is 0) while others have configured a rule not to bill all calls whose duration is less than a certain threshold. www.yo.co.ug On 8/13/08, Klaus Darilion [EMAIL PROTECTED] wrote: Steve Murphy wrote: On Tue, 2008-08-12 at 16:39 +0200, Klaus Darilion wrote: Hi! I wonder how Asterisk measures the call duration. The CDR files have a accuracy of seconds. Thus, what happens if the call duration is 0.3 seconds. What will Asterisk report? 0 seconds? 1 second? What logic will be used by Asterisk: floor? ceil? round? thanks klaus Klaus-- The duration/billsec fields are stored as simple integers. A simple integer subtraction is performed for both; duration is end time minus start time; billsec is end time minus answer time. Operations are done on system time, in seconds. If the .3 sec spans a system second increment, then the time will be 1, if not, then the time will be 0. It would seem to me the probability of .3 sec spanning a clock tick would be .3... CDR's do, internally, store finer increments than seconds. (struct timeval), but the interface yields plain seconds. I just checked the code, and sure enough, just the seconds field is used. So, truncation seems to be the rounding method. In general, we never fussed much about the microseconds, because on most interfaces, the slop in how much time it took to make a connection made the precision laughable. Hi Steve! Thanks for the detailed information. What about the following scenario: ANSWER and HANGUP happens in the same second. Thus, the call duration will be 0 seconds. How are such use cases usually handled in the billing system? Are you billing the user (e.g. 1 second or the minimum fee) if the call is ANSWERED even if Asterisk reports 0 seconds? regards klaus ___ -- 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 -- Sent from Gmail for mobile | mobile.google.com ___ -- 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] a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal)
Hi Everyone, Those of you who have a simple home-based Asterisk box might be interested in a simple Win32 (Win2K or WinXP) interface to the AMI manager. The quick-start versions merely require unzipping with NO Installation - hence, NO Uninstall (i.e., no registry writes at any time by the install nor by the program). (Unfortunately) the INSTALL version does write to the registry due to the database licensing requirements. Would suggest that you download the PDF and, if interested, (or if you hate to read manuals, just ), download the quick-start version which only requires 3 settings in Asterisk's manager.conf file (the user name, the password, and the read/write privileges - program defaults to the 5038 port). The program was really written as a nostalgic cruise down the old Pascal OOP thruway, and not as a contender to the likes of FOP, etc. Pascal has nice features such as declaring any of your functions inline; or for that matter writing inline Assembler code which was the language in the '70s (that is the 1900's, by the way). The Win2K version was compiled on an old Delphi 5 compiler (and for you young'uns, that was circa 1999 when Win2K was unveiled). However, fear not, the WinXP version was compiled with the latest Delphi 2007 R2. However, I did NOT insert some required Vista enabling statements (such as for the glass effect), since I have no interest in testing it (yet) in Vista; so, the XP version may or may not function well in XP compatibility mode within Vista. As for my Subject - Is anyone in this Asterisk group doing anything using Lazarus and FreePascal for the Asterisk box? The FreePascal compiler is a total (and, yes, an open source work in progress) cross-platform compiler. What I mean is, it can compile for Win, Mac, and Linux, but also for about half a dozen CPU's. The documentation for the compiler is an outstanding example for open-source projects. Downloads and info at: http://www.jerryh.us/Downloads/amifiles.htm Gerald Harshany, Ph.D. Professor Emeritus of Mathematics And again, for you young'uns, Emeritus simply means ancient :) ___ -- 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] Stupid Timeout Question
Hi, It may have to do with the version of Asterisk. I have (basically) the same coding on an Asterisk V1.4.18 box, and a V1.6 SVN test box - in both boxes the Asterisk does execute the = t,1,Playback(connection-timed-out) when nothing is entered. The only differences I can see between your coding and mine, is that a) I simply use the default timeout (i.e., WaitExten() ); but don't see why this matters, and b) I use the m option in the Background command, since I have a one-key extension. You could try using, exten = s,n,Set(TIMEOUT(absolute)=5) before the Background command, and see if this works. Gerald H. ___ -- 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] Asterisk V1.6.0 SVN debug WARNING(6830) a bug or deliberate?
Hi lists,
Does anyone know if the following error message (from a debug screen) was
a
deliberate change from the behavior in asterisk V1.4.18 or just an
overlooked
parsing error in progressing to V1.6.0? Since, in this case, the string (Hi
there)
is quoted, it doesn't seem as though the parser should take notice about
about the
interior of a 'word'. However, if it is deliberate, then so be it. (a yellow
NOTICE
would be more soothing than a red WARNING) :-)
Gerald Harshany
WARNING(6830): pbx.c:7557 pbx_builtin_setvar: Please avoid unnecessary
spaces on variables as it may lead to unexpected results
('DB(Knowselgreat/Hi there)' set to ' myfile ').
Using current Asterisk version: SVN-branch-1.6.0-r114304 (on Ubuntu) and
Zaptel current SVN 1.4
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Re: [asterisk-users] Roaming callback?
${FromPathFile} ${DestPathFile} 1/dev/null 21 );one long line
exten = s,n,Return()
[doringback] ; in a NEW CHANNEL NOW-the call-file created channel Zap/3
exten = s,1,Verbose(== in context doringback ready to dial ring back
caller)
; THE Zap/2 CALL WILL OCCUR (USUALLY) BEFORE THE CALL-FILE CALL
;
exten = s,n,Dial(Zap/2,20,r)
; check DIALSTATUS etc
exten = s,n,Hangup()
Hope this helps,
Gerald Harshany
Original Message
Subject: [SPAM] Re: [asterisk-users] Roaming callback?
From: SIP [EMAIL PROTECTED]
Date: Mon, April 28, 2008 1:25 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Jerry Harshany [EMAIL PROTECTED]
Jaap Winius wrote:
Quoting Jerry Harshany [EMAIL PROTECTED]:
There is an additional alternative for a ringback to a caller, which
is to use the Call File capability as noted in Van Meggelen's
Future of Telephone; 2nd ed, p306.
As it says in the book, call files allow calls to be created through
the Linux shell. If you've used this to create a roaming callback
service, then you must have created something that allows users to
submit a phone number to be called back on, after which a .call file
is created and moved to the /var/spool/asterisk/outgoing/ directory.
sleep 8s
mv $1 $2
exit 0
This looks like the step that moves the newly created call file to the
aforementioned directory.
In my case, when the caller calls in to 'asterisk', he is prompted
for the number he wishes to call. The caller can be at a US or
international number, and he can call any US or international
number, WITH or WITHOUT ringback. In other words the caller
designates whether this is a direct connect call, or a ringback (and
then bridge the called number). I have the complete flexibility of
my dial plan extensions to do as I wish with the phone numbers.
This is what I'm really interested in! How did you manage this? Would
you be willing to share how you did this?
I would imagine if it's a callback, it creates a callfile. If it's not,
it just connects the call as it would normally. We have a similar thing
for our business customers built using a reasonably simple agi script to
do verification of the caller/account and creation of the call files. A
rather simple Dial command can handle the direct connection after
verification, and a rather simple call file can handle the callback. The
hardest part was getting the DTMF reading to work well. ;)
N.
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Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk
Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better. The reason I asked is because IAX works better through firewalls and is easier to troubleshoot. It's not as widely deployed as SIP, but it does work around some major things that SIP makes harder. I'm not sure of the quality or lineage of the JAIN application code, so can't comment if it's a good jumping off point. I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Subscription is used for presence. It can be used in an IM type app, or to light up a button on a phone when someone is busy. It shouldn't be needed to exchange a call though, and if you can do it without the subscription piece then it could help to pin down the issue you are having. (It might be _just_ the subscribe that is having an issue). I should have time later this afternoon to check your traces, and I'll try and give Jain a kick. Thanks, Gerald. ___ --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] Can't create audio conversation betweensoftphonesthrough Asterisk
Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Glad to hear you were able to get some traction with the voice calling. Is the presence bit something that is critical to your custom app? I'm going to be fiddling with some soft phone stuff soon, so I am still planning on taking a peek at Jain just for the heck of it. Keep me updated on your progress, and if you need any assistance, give me a shout. Thanks, Gerald. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection
Hi Zeeshan, On 5/13/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I've solved this problem. It was very easy (only if I knew how to do it before). I changed the UDP ports, i.e. 1. In sip.conf, bindport=5070 2. In my IP Phone server settings, www.myserver.com:5070 Now it seems to be working good and I hope there'll be no more problem with it. Sorry for not replying earlier; I got your note late, and then when I woke up had no Internet. Ah, the joys of Rogers. I'm glad to hear you solved it -- my only concern would be if you now want to connect ordinary 5060 looking phones. I will do a bit of research, I'm sure Asterisk can bind to more then one port. Thanks, Gerald ___ --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] Any Suggestions for Election Polling Application?
Looking to set up an outbound only Asterisk installation for 5 to 10 attendants that will cold calling phone numbers in a database. The customer would like the server to call the numbers as needed and transfer the call to an open attendant if a voice response is detected. The customer called this call banking but it does not seem to translate directly into what Asterisk calls it? Would Asterisk be able to do this? Anybody have good experiences with softphone software? Would Asterisk able to tranfer the person's name/phone number back to the softphone once the connection is made? Any suggestions for SIP phones? Any trouble with using ITSP like Vonage if the user has a good internet connection? -- Regards -- Gerald Drouillard Technology Architect Drouillard Associates, Inc. http://www.Drouillard.ca ___ --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] Call Center with No TDM components
On Wed, 19 Apr 2006, Abhimanyu Rapria wrote: Transcoding and Recording is being done at VICIDIAL/ASTERISK Dialer and load average is 1.5 for 12 agents and pacing of 1.1 to 1.2 What is the average CPU utilization you observe with these load averages? Regards, Gerald. ___ --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] te110p and interrupts
Hi Paul, Thanks for the message! On Sun, 16 Apr 2006, Paul Hewlett wrote: [...] I am curious.. Have you tried disabling CPU1 by setting isolcpus=1 on the kernel command line ? This will make the kernel ignore the second CPU - you can then run asterisk on it by using the taskset command (from schedutils) taskset 0x0001 asterisk -p and asterisk wlll run on a CPU all on its own. I was about to try this and wondered if you might give it a try and report back. I haven't done this yet. Once we have physical access to the machine, I'll make sure we try this out and see what difference it makes. Cheers! Gerald. ___ --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] te110p and interrupts
On Mon, 17 Apr 2006, stoffell wrote: Interesting. Now 'why' do they suggest it, is it because older IO-APIC are 'broken' on some boards? I'm very curious as to 'why', [...] Most likely this is why. Regards, Gerald ___ --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 cards, so disappointing !
Hi Steve, Thank you for your very enlightening message! On Sat, 15 Apr 2006, Steve Underwood wrote: [...] modem it must be applied end to end by the modems themselves. The real killer, though, is imperfect timing. [...] and its not always always available within a PC. PCs are designed around best efforts handling of data. They don't handle continuous streaming of media well, even if the data rate is fairly low. They handle it especially badly if latency must be kept low, as is the case with I have come to understand and appreciate this fact more and more through painful experience. [...] That said, a well design PC environment can achieve the timing needed for FAX calls, as long as you don't load it up too much. In your opinion, short of re-engineering the PC, is there anything that can be done to step up the timing accuracy (and hence up the real-time performance) of the PC? What [hardware-based] technical action would you think can up the real-time performance of the PC? Regards, Gerald. ___ --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] te110p and interrupts
On Tue, 11 Apr 2006, Andrew Kohlsmith wrote: Please do not open your mouth to spout nonsense if you do not know what you're talking about. [...] Again, if the IO-APIC is reporting that the card is on its own IRQ, it really, truly, honestly *IS* on its own IRQ. The reason that it is suggested to disable the IO-APIC is that on many low-end systems, the IO-APIC is plain old broken and causing other issues. I don't think I've run across a system board in the last year or two with that issue, though. It's always been on older P3 and early P4 systems. Allow me to comment that Digium actually recommends turning off APIC and using lspci -vb to troubleshoot this kind of shared-interrupt problem. Cheers, Gerald. ___ --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] te110p and interrupts
Hi, I've been battling with a similar issue: a) I wrote a script to periodically run the command cat /proc/interrupts and figure out the interrupts per second. I run this script for over 24 hours and never once did the difference between the preceeding and succeeding interrupt counts go below 1005 (wierd result because of (b) below); b) zttest was reporting very bad results; c) lspci -vb was reporting that the TE110P shared an IRQ with the Gigabit Ethernet Card (IRQ 11) d) lspci -vv was reporting that the TE110P was on an IRQ of its own (IRQ 24) probably because of APIC (wierd because of (c) above); e) Users reported intermittent bad audio; Below are the [experimental] steps I took: a) I'm running a Dual 3.2 GHz machine - the network card is services by CPU0 - I set the smp_affinity value for the Digium card to be CPU1 b) I disabled the userland 'irqbalance' process which keeps switching the Digium card between the CPUs c) I increased the PCI LATENCY_TIMER value for the TE110P to a value higher than the Gigabit Card. So far, things are looking quite good - zttest is reporting very encouraging worst-case figures when run over a period of over an hour (it reports 99.98% worst case at off peak time and 99.77% when run during the busy hour). Ultimately when I have physical access to the machine, I will change the PCI slots to see if getting lspci -vb to report that the card is on its own IRQ will improve performance further. Cheers, Gerald. ___ --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] get no connection, very often, but not allways, why?
Hi, I have an ISDN phone connected to a hfc-s card. I use it to phone via an iax provider to foreign countries. Inside my country it works reliable, but to other country it happens very often that the other side hears ringing and before it can take the phone the line is dropped. What makes me wonder is that I hear no ringing at all. With asterisk -c I get this: Asterisk Ready. *CLI == Primary D-Channel on span 1 up for TEI 64 -- Accepting overlap voice call from '' to 'unspecified' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, IAX2/user:password@sip.coco-connect.de/XXX) in new stack -- Called user:password@sip.coco-connect.de/XXX -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 -- Channel 0/2, span 1 got hangup -- Hungup 'IAX2/62.180.50.221:4569/1' == Spawn extension (extern, XXX, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Does the line -- Channel 0/2, span 1 got hangup mean that the ISDN-Phone drops the line first? If yes, could it be, because the phone gets no ringtone a too long time? I use no timeout for this channel. Thanks for any help. Regards Gerald ___ --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: ***SPAM*** Re: [Asterisk-Users] D-Link DVG-1402S
On Sun, 05 Mar 2006 19:56:13 +0800 Stephen Arulraj [EMAIL PROTECTED] wrote: Come on.! Don't tell me no one has ever had a problem on this model with asterisk? Live it up guys... and make a few comments maybe you would get more answers if you wouldn't steal a thread, but would create your own. For me it is not clear how your message belongs to the thread No audio on PRI. Gerald ___ --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] suddenly iax calls don't work anymore
Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Gerald ___ --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] suddenly iax calls don't work anymore
Thanks, that helped Gerald On Wed, 04 Jan 2006 14:39:51 + Faris Raouf [EMAIL PROTECTED] wrote: Gerald Dachs wrote: Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Gerald Someone will probably correct me, but it looks like you are trying to use the g729 codec for your calls (or coco-connect.de is forcing you to use g729), but this requires a license from Digium and is not installed on your machine. Try using a different codec if possible or, if you do have a g729 license try re-installing the codec and re-activating it. I think this may solve the problem. But as I say, someone may correct me - I may be completely wrong about this. Faris. ___ --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] Asterisk 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have you by chance set immediate to yes? IIRC, there's a feature that will send you to the configured context as soon as you pick up your phone (this is in zapata.conf). Might be worth checking that out. Cheers, Gerald. ___ --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 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Remco Barende wrote: I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes What I actually meant is that you should turn this off if you don't need the functionality. Most likely you are defining the extension channel after the phone line thus it is inheriting the setting as well. Gerald. ___ --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] has someone zaphfc with xenomai working?
Hi, build worked just fine, had only to change rt_get_time to rt_get_time_ns, according to the xenomai guys this is the same in xenomai. After loading the zaphfc and the realtime modules the realtime interrupts increase. The hfc-s card is found and everything shows up fine in /proc/zaptel/1, but zttest does not show any throughput. Does someone work with this configuration? Regards Gerald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [patch] sqlite3 support for asterisk 1.2.0
Hi,
I changed cdr_sqlite so that it builds with sqlite3. I named the new module
cdr_sqlite3. It builds, but I will not be able to test it the next days.
I provide it anyway, maybe a brave heart gives me response.
Gerald
diff -Nur asterisk-1.2.0.orig/cdr/cdr_sqlite3.c
asterisk-1.2.0.sqlite3/cdr/cdr_sqlite3.c
--- asterisk-1.2.0.orig/cdr/cdr_sqlite3.c 1970-01-01 01:00:00.0
+0100
+++ asterisk-1.2.0.sqlite3/cdr/cdr_sqlite3.c2005-11-23 14:01:29.0
+0100
@@ -0,0 +1,244 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2004 - 2005, Holger Schurig
+ *
+ *
+ * Ideas taken from other cdr_*.c files
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ *
+ * Changes for SQLite 3 by Gerald Dachs
+ *
+ */
+
+/*! \file
+ *
+ * \brief Store CDR records in a SQLite database.
+ *
+ * \author Holger Schurig [EMAIL PROTECTED]
+ *
+ * See also
+ * \arg \ref Config_cdr
+ * \arg http://www.sqlite.org/
+ *
+ * Creates the database and table on-the-fly
+ * \ingroup cdr_drivers
+ */
+
+#include sys/types.h
+
+#include stdio.h
+#include unistd.h
+#include string.h
+#include stdlib.h
+#include sqlite3.h
+
+#include asterisk.h
+
+ASTERISK_FILE_VERSION(__FILE__, $Revision: 1.11 $)
+
+#include asterisk/channel.h
+#include asterisk/module.h
+#include asterisk/logger.h
+#include asterisk/utils.h
+
+#define LOG_UNIQUEID 0
+#define LOG_USERFIELD 0
+
+/* When you change the DATE_FORMAT, be sure to change the CHAR(19) below
to something else */
+#define DATE_FORMAT %Y-%m-%d %T
+
+static char *desc = SQLite3 CDR Backend;
+static char *name = sqlite3;
+static sqlite3* db = NULL;
+
+AST_MUTEX_DEFINE_STATIC(sqlite3_lock);
+
+/*! \brief SQL table format */
+static char sql_create_table[] = CREATE TABLE cdr (
+ AcctId INTEGER PRIMARY KEY,
+ clidVARCHAR(80),
+ src VARCHAR(80),
+ dst VARCHAR(80),
+ dcontextVARCHAR(80),
+ channel VARCHAR(80),
+ dstchannel VARCHAR(80),
+ lastapp VARCHAR(80),
+ lastdataVARCHAR(80),
+ start CHAR(19),
+ answer CHAR(19),
+ end CHAR(19),
+ durationINTEGER,
+ billsec INTEGER,
+ disposition INTEGER,
+ amaflagsINTEGER,
+ accountcode VARCHAR(20)
+#if LOG_UNIQUEID
+ ,uniqueid VARCHAR(32)
+#endif
+#if LOG_USERFIELD
+ ,userfield VARCHAR(255)
+#endif
+);;
+
+static int sqlite3_log(struct ast_cdr *cdr)
+{
+ int res = 0;
+ char *zErr = 0;
+ struct tm tm;
+ time_t t;
+ char startstr[80], answerstr[80], endstr[80];
+ int count;
+ char *sqlstmt;
+
+ ast_mutex_lock(sqlite3_lock);
+
+ t = cdr-start.tv_sec;
+ localtime_r(t, tm);
+ strftime(startstr, sizeof(startstr), DATE_FORMAT, tm);
+
+ t = cdr-answer.tv_sec;
+ localtime_r(t, tm);
+ strftime(answerstr, sizeof(answerstr), DATE_FORMAT, tm);
+
+ t = cdr-end.tv_sec;
+ localtime_r(t, tm);
+ strftime(endstr, sizeof(endstr), DATE_FORMAT, tm);
+
+ for(count=0; count5; count++) {
+ if ((sqlstmt = sqlite3_mprintf(
+ INSERT INTO cdr (
+ clid,src,dst,dcontext,
+ channel,dstchannel,lastapp,lastdata,
+ start,answer,end,
+ duration,billsec,disposition,amaflags,
+ accountcode
+# if LOG_UNIQUEID
+ ,uniqueid
+# endif
+# if LOG_USERFIELD
+ ,userfield
+# endif
+ ) VALUES (
+ '%q', '%q', '%q', '%q',
+ '%q', '%q', '%q', '%q',
+ '%q', '%q', '%q',
+ %d, %d, %d, %d,
+ '%q'
+# if LOG_UNIQUEID
+ ,'%q'
+# endif
+# if LOG_USERFIELD
+ ,'%q'
+# endif
+ ),cdr-clid, cdr-src, cdr-dst, cdr-dcontext,
+ cdr-channel, cdr-dstchannel, cdr-lastapp,
cdr-lastdata,
+ startstr, answerstr, endstr,
+ cdr-duration, cdr-billsec
[Asterisk-Users] zaphfc not generally compatible with kernels = 2.6.13
Hi, I am very new to asterisk so forgive me if I tell something stupid. I am investigating currently a problem with zaphfc. I get only very few interrupts, they don't get lost, the interrupt count increases only very slowly. I really don't know where to look for the problem, so I looked here and there and found the following line in zaphfc.c from bristuff-0.2.0-RC8o: schedule_timeout((30 * HZ) / 1000); // wait 30 ms IIRC the default HZ in 2.6.13 (or was it 2.6.14?) is 250. In our kernel HZ is 100. So the wait gets too short, should the driver not check that CONFIG_HZ_1000 is set? I am not sure that this is the reason for my problem, but I build currently a new kernel and will test it. Gerald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc not generally compatible with kernels = 2.6.13
On Mon, 7 Nov 2005 23:06:24 +0100 Gerald Dachs [EMAIL PROTECTED] wrote: Hi, I am very new to asterisk so forgive me if I tell something stupid. It has happend, my post was stupid I am investigating currently a problem with zaphfc. I get only very few interrupts, they don't get lost, the interrupt count increases only very slowly. I really don't know where to look for the problem, so I looked here and there and found the following line in zaphfc.c from bristuff-0.2.0-RC8o: schedule_timeout((30 * HZ) / 1000); // wait 30 ms schedule_timeout is a kernel call that gets ticks as arg and not ms. Gerald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't build Asterisk on SuSE
SuSE Linux Enterprise Server 9 Asterisk 1.2.0 beta1 I am trying to build and install Asterisk on SuSE. I started with a fresh full installation of SuSE. The last lines of stdout and the full stderr are attached below. Thanks very much for your assistance. -Ramon F Herrera [cutted much lines] res_crypto.c:15:25: openssl/ssl.h: No such file or directory res_crypto.c:16:25: openssl/err.h: No such file or directory res_crypto.c:75: error: parse error before RSA [cutted much lines] This is my first post to this list, I have no experiences with asterisk, but this problem is an easy one and it is not asterisk related. The problem is that you didn' t read the error messages. In the lines above you can see that you did't install the development files for openssl. I don't know how this rpm is named in suse, but in my distro it is called openssl-devel. Gerald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Application: Broadcast
Hi Steve, On Thu, 13 Oct 2005, Steve Daniels wrote: What excatly does it do? What messages does it send out? And what software needs to be configured to listen for these messages? Bret explained mostly what the software does in a basic use case where you would like a nice window to pop up with say the caller id details of an incoming call. With this same software, you may selectively broadcast messages for example, you may only want the sales crew to see information about a given caller and not other groups. For example: [sales-context] exten = s,1,Answer exten = s,2,Broadcast(This is a sales call|group=sales) exten = s,3,Dial(whatever) In such a case, you will need to have configured the sales computers with a group attribute set to sales for example: [192.168.1.1] port = 10296 group = sales [192.168.1.2] port = 10345 group = sales [192.168.1.3] port = 19002 group = technical In such a case as above, onlye the first two machines (192.168.1.1 and 192.168.1.2) will be notified. All you need configured on the machines that need to receive these messages is software like YAC (Yet Another Callerid program) which you may get from http://sunflowerhead.com/software/yac/ You will only need to configure the broadcast application to connect to the right port. The usage and testing informtion is quite well documented in the accompanying README file. Hope you find it useful! Cheers, Gerald. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Application: Broadcast
Hello, I've released an Asterisk application under the terms of the GNU GPL. You may find it here: http://psg.com/~begg/projects/ A short exerpt from the README: -- Broadcast is an Asterisk (http://www.asterisk.org) application which you may use to send a generic message over TCP/IP to any number of computers running software configured to listen for these types of messages. Being written in C, Broadcast will be dynamically loaded onto the Asterisk program on startup, making it a highly reliable and scalable option when compared with other solutions based on the Asterisk Gateway Interface (AGI) system... -- Hope someone finds it useful! Cheers, Gerald. PS: Sorry for the cross posts! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Getting phpconfig to work?
Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; It's quite likely that your Apache+PHP installation is incomplete / broken. You may want to check that out. Cheers, Gerald. ___ 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] wctdm and two tdm cards
If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. To completely reset the TDM cards before they can be reliably detected again, you may have to completely power down the machine - even to the extent of pulling out the power plug and replacing it, then booting up. Regards, Gerald. ___ 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 Card Problems
Hi Mark, On Mon, 28 Feb 2005, Mark Kidd wrote: modprobe zaptel - no problems [EMAIL PROTECTED] root]# modprobe wcfxo I'm just curious, did 'modprobe wcfxo' ever work? I seem to remember that for the TDM400P suite, the module to load was (rather confusingly) 'wcfxs', even though you've got FXO modules on the card. we are running the 4 port fxo digium card. so normal the modprobe wcfxs no problems modules load and board comes up after starting asterisk. That's TDM04B, right? If you don't have the Wildcard X100P (or something of the sort) plugged too then I see no reason to be loading 'wcfxo'. Hope that helps. Regards, Gerald. PS: The module name was later changed from 'wcfxs' to 'wctdm' (to avoid confusion I think. So, if you have no X100P, I think you can safely ignore loading 'wcfxo') ___ 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] IPCB
On Thu, 24 Feb 2005, HASS, JOHN wrote: type=peer For some reason type=friend seemed to solve a similar problem I had (not with IP Clearing Board, though). I was kinda too busy to figure out why it solved the problem, actually [sorry] but it *may* be worth checking out. Then, just to clarify, that section in your sip.conf seems to suggest that you've only configured your server to allow calls to be terminated *from* IPCB? I.e the IPCB registering with your Asterisk server. Perhaps you might want to think of a register statement? I admit am not completely familiar with the way they get things running though, just a guess. Hope that helps. Regards, Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to manage Digium TDM04B outgoing calls correctly
So if you think the server can handle 5 TDM400P cards let me know. I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12 analog phones. There are no outstanding issues that havent been solved by tweaking a particular config option (e.g echo, callprogress issues etc...). Gerald. ___ 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] TE410P card in an HP-Compaq DL380 G4 server
Yup, I found their support very unhelpful and unwilling to go the extra (or even the first) mile.. Might ACPI (not APIC) have anything to do with this condition? I once had a hard time with a bunch of cards which were not taking interrupts. I disabled ACPI interrupt routing (from the grub boot prompt, put pci=noacpi) and everything started working. Well, these were TDM400P cards (5 of them) anyway with a different type of machine altogether but it just might be worth checking out. Rgds, Gerald. ___ 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] Multiline / Console / Receptionist phone
I have been looking to see if this type of phone can be implimented in *. I have found nothing conclusive. Is any out there using a multiline / mutlifunction phone typically used by a receptionist for transfering / routing calls? I need to know how this is accomplished or what alternative exists for this. Thanx! Gary P. -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] Multiline / Console / Receptionist phone
Does this phone have LEDs showing lines in-use? Thanx! Gary P. Tracy R Reed wrote: On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly: I have been looking to see if this type of phone can be implimented in *. I have found nothing conclusive. Is any out there using a multiline / mutlifunction phone typically used by a receptionist for transfering / routing calls? I need to know how this is accomplished or what alternative exists for this. I am using the Snom 220 with the hint extension priority with success. ___ 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 -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] Granstream phones message button
Using: Software Version: Program--1.0.5.16 Bootloader--1.0.0.21 HTML--1.0.0.41 VOC--1.0.0.6 I will pursue changin firmware. Is this difficult? Gary P. Derek Conniffe wrote: I meant to say "downgrade to 1.0.5.11 Derek Derek Conniffe wrote: It sounds like you might be using the broken BT firmware - are you using 1.0.5.16? If you are you'll find that the message button and the auto-dial feature do not work. You can downgrade to 1.0.5.16 but I've read that the newer [BETA?] 1.0.5.18 works too. Derek Gerald J. Puhl wrote: To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to go right to voicemail. Gary P. -- * Prototypes * * Patterns * * Models * * Dies * * Fixtures * * Please visit /www.jppattern.com http://www.jppattern.com// for more information about J.P. Pattern, Inc. * ___ 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 -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] Ripping CD audio for MOH
Tom: I downloaded this freeware @ http://www.eusing.com/ (free CD to MP3). It converted my CD (CDA tracks) to MP3. Copied them onto my * server mohmp3 dir. You will need mpg123 for moh. I downloaded the rpm and installed. Everthing worked the first time I set it up. Gary P. Thomas Johnson wrote: Hello- I've got some audio CDs that I'd like to use for MOH. What's the best way to do this? I don't care if it's mp3 or some other format - whatever will work best. What applications (osx or linux) are best? Optimal settings? Thanks- Tom ___ 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 -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] Granstream phones message button
To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to go right to voicemail. Gary P. -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] SoftPhone on * with X-Lite or iaxComm (1 X100P card)
To all: I am having an echo problem with X-Lite and iaxComm. I am using the monitor speakers and a desktop microphone. My problem is that the sound from the speakers is repeated by the microphone causing an echo that is annoying. Is this correctable? I have searched throught this archives and played with various settings but have not been able to fix this. I have purchased a couple if different microphones that I thought may cure this problem, but nothing changed. Is this just simply a bad chioce and I need to use a headset rather than an open speaker microphone setup? Thanx, Gary P. PS: I am an * newbie, please go easy on me. -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] X101P interface (asterisk newbie)
To all: I am researching the feasibility of replacing our current PBX (ATT Partner Plus) with an * PBX. I have purchased an X101P card and I have * running of a FC2 machine. The X101P is connected to an extension on our current PBX. Many archives exists regarding X101P cards but I just need some guidance. First the X101P did not detect a ring, so I found an email where someone reconfigured the MINPEGTIME by decreasing the amount (did that and it seamed to work). Now sometimes it seams to get stuck and not detect a ring unless is repeatedly stop and restart *. Output form * in CLI shows a hang-up: Hang-up: channel: 1 index = 0, normal = 12, callwait = -1, thirdcall = -1 disabled echo cancellation on channel 1 Set option TDD MODE, value: OFF(0) on Zap/1-1 Updated conferencing on 1, with 0 conference users -- Hungup 'Zap/1-1' I assume that this means I am not having a trouble detecting a hangup? Is the X101P a finicky card (and perhaps a bad choice)? Or am I just not setting things right (and stupid). -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ 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] TDM31B has no interrupts?
Hi, I've installed a TDM31B card successfully but had a few problems making calls through it - summary is below: o Calls cannot be placed using an analog phone o The interrupts count value in /proc/interrupts remains at zero (see below) CPU0 0: 7495 XT-PIC timer 1: 7 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 2 XT-PIC rtc 9: 0 XT-PIC acpi 11:508 XT-PIC eth0 12: 0 -- XT-PIC wctdm -- 14: 2662 XT-PIC ide0 NMI: 0 LOC: 0 ERR: 0 MIS: 0 o I've tried this card in all three PCI slots but no luck o I've tried two other TDM31Bs in a similar manner with no luck o I've tried the same with a TDM22B and get similar behaviour Could all my PCI slots be dead or is it likely that all 3 TDM31B cards are dead + the TDM22B? Any clues are highly appreciate. Rgds, Gerald ___ 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] TDM31B has no interrupts?
Hi, Thanks for taking time to answer. Not enough info in the above to hint at the problem. What linux distro, SuSE Linux 8.2 2.4.20-4GB what does your /etc/zaptel look like, For the TDM22B card: fxoks=1-2 fxsks=3-4 loadzone = uk defaultzone=uk zapata.conf signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callprogress=yes busydetect=1 busycount=7 relaxdtmf=yes channel = 1,2 signalling=fxs_ks group=2 context=incoming channel= 3,4 , what steps did you take to start the drivers (eg, modprobe, ztcfg), etc. linux:~ # modprobe zaptel Warning: loading /lib/modules/2.4.20-4GB/misc/zaptel.o will taint the kernel: no license See http://www.tux.org/lkml/#export-tainted for information about tainted modules Module zaptel loaded, with warnings linux:~ # modprobe wctdm Warning: loading /lib/modules/2.4.20-4GB/misc/wctdm.o will taint the kernel: no license See http://www.tux.org/lkml/#export-tainted for information about tainted modules Module wctdm loaded, with warnings linux:~ # ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. linux:~ # Since we're forced to guess at that stuff, here's a list of things that might have an impact. - ensure the definitions in /etc/zaptel.conf are reasonable Hope they are. - run ztcfg -vvv from the command line. Any errors? No errors as above. - run 'modprobe wctdm', any errors? The only message that shows is a warning about the kernel being tainted. I checked the link I was referred to and it should really have no effect on the operation of the card (s). - what does dmesg, lspci, and lsmod output say? dmesg: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) Registered tone zone 4 (United Kingdom) Registered tone zone 4 (United Kingdom) lspci: 00:00.0 Host bridge: VIA Technologies, Inc. P4M266 Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP] 00:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus Master IDE (rev 06) 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97 Audio Controller (rev 50) 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) 01:00.0 VGA compatible controller: S3 Inc. [ProSavageDDR K4M266] lsmod: Module Size Used byTainted: P wctdm 25568 0 (unused) zaptel183616 0 [wctdm] snd-pcm-oss45888 0 (autoclean) snd-mixer-oss 13560 0 (autoclean) [snd-pcm-oss] isa-pnp29672 0 (unused) ipv6 134388 -1 (autoclean) raw139414516 0 (unused) ieee1394 32880 0 [raw1394] via-rhine 12176 1 mii 2304 0 [via-rhine] snd-via823312516 0 snd-pcm62912 0 [snd-pcm-oss snd-via8233] snd-timer 11904 0 [snd-pcm] snd-ac97-codec 31152 0 [snd-via8233] snd-mpu401-uart 3360 0 [snd-via8233] snd-rawmidi13824 0 [snd-mpu401-uart] snd-seq-device 4000 0 [snd-rawmidi] snd35940 0 [snd-pcm-oss snd-mixer-oss snd-via8233 snd-pcm snd-timer snd-ac97-codec snd-mpu401-uart snd-rawmidi snd-seq-device] soundcore 3396 0 [snd] reiserfs 200532 1 - what does zttool show? I realized I need the package libnewt to get this to compile. Meanwhile does the above information reveal anything? Is there any BIOS setting I need to tweak? Thanks in advance. Rgds, Gerald. ___ 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] TDM31B Interrupt Issue SOLVED! :-)
-- My apologies if this gets posted twice. I made a mistake with my from address. -- -- Hi All, Many thanks to everyone that gave input on the above issue. I'm glad to announce its been solved. The trick: -- TURN OFF ACPI! -- With SuSE you can do this by setting the boot option pci=noacpi. Everything now works flawlessly except for some suspicious static that I heard on one of the 3 TDM31B cards, which vanished after I reloaded the modules. Incidentally the technical reference booklet that came with the PC says the slots are PCI ver 2.1 compliant. I almost thought that would be a problem. Well, it turns out it isnt after all. Thanks again. In case anyone is interested, I've included my scratch notes on what I went through with this. Rgds, Gerald. --[Use at your own risk!]-- - Download asterisk, zaptel from CVS - Hack zaptel.c and wctdm.c modules to have the kernel_version string in them. This will allow them to be loaded (had to do this for SuSE :-(). - Edit /etc/zaptel.conf to tell the signalling, zone etc... - Edit /etc/asterisk/zapata.conf to reiterate this stuff for asterisk - Run ztcfg -vvv and note the output HARDWARE WOES - Disable xwindows - Disable USB (remove /etc/hotplug/usb.rc or rename it) - Tweak BIOS IRQ stuff THE PCI SLOTS SHOULD BE PCI 2.2 COMPLIANT! TURN OFF ACPI I.E WHEN BOOTING USE pci=noacpi -- ___ 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] 3 - TDM31B Card Installation Difficulty
Hi, o I purchased 3 TDM31B cards and fixed them in my computer (in 3 PCI slots) o I downloaded the latest Zaptel source from CVS, compiled it and loaded modules zaptel.o and wctdm.o. o I successfully configured them from /etc/zaptel.conf as shown in the information below. ztcfg returned no errors - see the report below. o I successfully configured /etc/asterisk/zapata.conf (see info below). o I configured an X-Lite phone to test with an analog phone plugged into one of the channels. The problems are: o I cannot make a call from the analog phone (Saachi phone, KX-T3223) connected to one of the FXS ports. When I pick up the receiver, I hear the dialtone but when I press the buttons, asterisk seems not to get the numbers dialled, both using pulse and touch tone dialling. o I can call the analog phone from X-Lite however on receiving, I cannot hear much voice. What I hear is choppy sound corresponding to whatever I say from the analog side. When someone speaks from the X-lite side, nothing is heard from the analog phone. o There are three FXS ports where there is no dialtone - but the phone is actually powered - I can hear touch tone / pulse when I dial. o There are three FXS ports that give neither power nor dialtone. What could the problem be? Any help will be highly appreciated. Please find below abit of information I thought may be useful. Please let me know if more is needed. EXTRA INFORMATION - linux:/usr/src/new # uname -a Linux linux 2.4.20-4GB #1 Mon Mar 17 17:54:44 UTC 2003 i686 unknown unknown GNU/Linux linux:/usr/src/new # linux:/usr/src/new # ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXO Loopstart (Default) (Slaves: 03) Channel 04: FXS Loopstart (Default) (Slaves: 04) Channel 05: FXO Loopstart (Default) (Slaves: 05) Channel 06: FXO Loopstart (Default) (Slaves: 06) Channel 07: FXO Loopstart (Default) (Slaves: 07) Channel 08: FXS Loopstart (Default) (Slaves: 08) Channel 09: FXO Loopstart (Default) (Slaves: 09) Channel 10: FXO Loopstart (Default) (Slaves: 10) Channel 11: FXO Loopstart (Default) (Slaves: 11) Channel 12: FXS Loopstart (Default) (Slaves: 12) 12 channels configured. linux:/usr/src/new # /etc/zaptel.conf: fxols=1-3 fxols=5-7 fxols=9-11 fxsls=4,8,12 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [channels] signalling=fxo_ls echocancel=16 echocancelwhenbridged=yes is in milliseconds pulsedial=yes group=1 context=default callprogress=yes busydetect=1 busycount=7 relaxdtmf=yes channel = 9-11 channel = 1-3 channel = 5-7 signalling=fxs_ls group=2 context=incoming channel= 4,8,12 Gerald. ___ 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] 3 - TDM31B Card Installation Difficulty
Hi Steve, If you call from X-Lite to the demo menus can you hear them clearly (no choppy sound)? Actually I can't - the sound is still choppy! Interesting. When I unload the zaptel and wctdm modules the problem goes away (I can hear the demo files quite clearly from the X-Lite phone). Given the problems you are having this might point to a bad TDM100P card. Mmm. I have a spare one. I'll replace the one that doesn't give dialtone and see what happens. Thanks alot Steve. I'll fix the card and let you know what happens. Rgds, Gerald. ___ 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] Dialing Out through Provider with Authentication
On Thu, 9 Sep 2004, Karl Brose wrote: In order to dial out to a sip provider, you need to configure that provider in your sip.conf file as a peer with your proper username and secret, etc. Cool! Just found that in the handbook too a second or two ago :-) Thanks for taking time to answer this. Three Cheers! Gerald ___ 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] Dialing Out through Provider with Authentication
On Fri, 10 Sep 2004, Johannes Hollerer wrote:
I tried to make a call to extension 2001 with the setting
[EMAIL PROTECTED] (Detailed: exten =
_7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1}) which does not work at
all - i always get the failure message: No such host
provider.com/2001 (the number i dialed) - why ??
What I understood from Karl's message is that you need to create a peer in
sip.conf. For example below:
-- sip.conf --
[myprovider]
type=peer
username=USERNAME
host=PROVIDER.COM
secret=SECRET
--
Then in extensions.conf, do the following:
--
exten = _7.,2,Dial(SIP/myprovider/${EXTEN:1})
--
This should work. What Karl meant is that using the statement below:
--
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED])
--
Will only work if you are dialilng a *specific* extension on provider.com.
The statement below:
--
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
--
Is illegal.
Cheers,
Gerald
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Re: [Asterisk-Users] Asterisk testbed for teaching connecting to a PRI-ISDN
On Fri, 10 Sep 2004, Francesco Delfino wrote: [...]One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. As more experienced people prepare to reply, I'd like to give my [highly theoretical] opinion (I'm still waiting for hardware I ordered): I think it is possible to just cross connect them, as long as you get the signaling right. In my opinion, the Box simulating the telco should signal as the network side and the one representing the company should signal as the customer side... Hope that makes sense. Cheers, Gerald. ___ 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] Dialing Out through Provider with Authentication
On Thu, 9 Sep 2004, Johannes Hollerer wrote:
I try to dial out through a Provider, but for that i need to be
authenticated - it actually does not work !.
For my tests I did not need to be authenticated. This is what I used in
asterisk:
exten = _7.,2,Dial(SIP/PROVIDER.COM/${EXTEN:1})
When I tried to use your scenario, as below
exten = _7.,2,Dial(SIP/USERNAME:[EMAIL PROTECTED]/${EXTEN:1})
Here's what I get in my logs:
Sep 9 18:10:56 WARNING[137570304]: chan_sip.c:902 create_addr: No such
host: PROVIDER.COM/72312
What I gather from this is that its not legal to Dial() like that. In my
limited SIP knowledge, it makes sense - you do not need to have a username
and / or password to place calls to extensions that a given provider (e.g
PROVIDER.COM) serves - if they do not serve those extensions, they will
give a 404 Not Found error.
Hope that helps...
Gerald.
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Re: [Asterisk-Users] Dialing Out through Provider with Authentication
But the provider also has a gateway to provide the possibility to
call to the pstn (and the pstn number exists) - so what i tried to
achive is to call an external pstn number thru that gateway. This
works if i connect the xlite client directly to the provider - then
i can dial the external number.
Alright, I see what you mean. Have you entered a register statement in
sip.conf, then? I.e something like
register = USER:[EMAIL PROTECTED]/EXTENSION
What I understand is that this will result in your Asterisk Server
registering on that provider's server as one of its users. Now the
question is how you dial out through this registration... Ya? That much I
don't claim to know. I just hope guys who have done this are reading this
thread. I'd like to learn this too.
However try using the dial below with the above register statement in
place (don't forget to reload your Asterisk server).
exten = _7.,2,Dial(SIP/PROVIDER.COM/${EXTEN:1})
Ideas, anyone else?
Gerald
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Re: [Asterisk-Users] Placing Asterisk between existing PBX and PSTN
Hi All, Just wondering if anyone could have by chance taken a look at the scenario below... I checked up http://www.asteriskpbx.com/index.php?menu=support and it looks like Asterisk-Users is the correct list to post this (I think...). I'd really appreciate any insight. Gerald. On Mon, 6 Sep 2004, Begumisa Gerald M wrote: Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup --- PSTN +---+ --|| ||--A1 --|| PBX ||--A1 --|| ||--A1 --|| ||--A1 +---+ A1,A2,A3,A4 are analog extensions Setup With Asterisk --- PSTN +--+ +---+ --||| |||| ||--A1 --|FXO Card|| Asterisk ||FXS Card|| PBX ||--A2 --||| |||| ||--A3 --||| |||| ||--A4 +--+ +---+ So they only pay for the Asterisk box, the FXO Card (TDM04B) and FXS Card (TDM40B). I'd appreciate any yes/no/been there answers. I just want to make sure about this, in case there's anyone that's done this before. Thanks in advance. Gerald. ___ 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] Placing Asterisk between existing PBX and PSTN
Hi Jens, Thanks alot for your input, I do appreciate it! [...] I would like to suggest that you don't try this with analogue lines (fxo) and extensions (fxs) - you will not be able to monitor call progress and lose all (possible) DDI information. Imagine my original setup was purely analog i.e I have 4 analog lines from the local telecom company. If I plugged these lines into the Asterisk FXO card and then plugged the PBX into the Asterisk FXS card, I'm thinking I would be able to use the calling card in addition to making VoIP calls application if for example I set up an extension context as below: [analogextensions] exten = 101,1,CallingCard exten = 101,1,Congestion exten = 102,1,Dial(SIP/[EMAIL PROTECTED]) I.e if anyone on the analog phones dials 101, they get the Calling Card application, if they dial 102, they get connected to some SIP phone somewhere etc... Would this minimum functionality work? Thanks! Gerald. ___ 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] Placing Asterisk between existing PBX and PSTN
Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup --- PSTN +---+ --|| ||--A1 --|| PBX ||--A1 --|| ||--A1 --|| ||--A1 +---+ A1,A2,A3,A4 are analog extensions Setup With Asterisk --- PSTN +--+ +---+ --||| |||| ||--A1 --|FXO Card|| Asterisk ||FXS Card|| PBX ||--A2 --||| |||| ||--A3 --||| |||| ||--A4 +--+ +---+ So they only pay for the Asterisk box, the FXO Card (TDM04B) and FXS Card (TDM40B). I'd appreciate any yes/no/been there answers. I just want to make sure about this, in case there's anyone that's done this before. Thanks in advance. Gerald. ___ 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] Re: Help with GPL license of Asterisk
On Thu, 02 Oct 2003 11:26:56 -0700, Jan Rychter [EMAIL PROTECTED] wrote: Having worked with GPL software quite a bit, also in the commercial world, and having gotten some legal advice, I believe that the anti-patent clauses in the GPL and LGPL are quite possibly the biggest problem preventing the use of GPL'd software by commercial entities, much bigger than the pass on the source and the rights requirement. Not really. Certainly it hasn't stopped lots of companies big and small from releasing GPL software. As I understand it (and as my legal counsel advises me) this effectively means that if I distribute GPL/LGPL code, I have to make sure that its distribution and re-distribution is not restricted by patents (or other restrictions). No, simply because that would be impossible (both because you would never be able to program given the number of patents you would have to search, and because it is entirely probable that no software is entirely patent free). What you can't do is knowingly license some source code/software under the GPL/LGPL if you are already aware of any patent or other issues that would in any way conflict with the redistribution of that code. If the code in question contains parts which some patents lay claim to, restricting distribution, then I must not distribute the code at all. Correct. The GPL/LGPL allow no further restrictions other than that of the GPL/LGPL itself. It is needless to mention that it is impossible to me to verify that no patents (worldwide!) lay claim to the code I'm distributing and impose restrictions upon its distribution. Sooner or later I'm going to find out that I do not comply with the GPL, because I distribute GPLd code even though there are patent restrictions that apply to it. Possibly. But those same issues apply to any software whether open source or closed source. Regardless of the license used patents would still apply, and would be enough to force you to stop distributing your software without an appropriate license (and possibly fee). An example of a particularly clear case of this problem is the XviD code (http://www.xvid.org/), which is GPL-licensed. It seems to me that the authors (copyright holders, to be precise) may distribute the software under any license they choose, but nobody else is allowed to re-distribute it, because they would be violating section 7 of the GPL, as the MPEG-4 compression is (in some countries) covered by patents requiring royalties to be paid. Wrong. The authors of xvid cannot license it under the GPL/LGPL because MPEG-4 is known to have patent license issues. In other words the patent issues place a restriction on distribution that violates the GPL, hence it cannot be GPL. This is not unique to xvid, the same issue applies to any of the mp3 decoders (like xmms) which cannot be GPL/LPGL licensed which is why at the very least Red Hat has removed those programs from their distribution. If those authors want to release open source codecs then they need to either: a) use another open source license that does allow restrictions on further redistribution (I believe the BSD license falls into this category but I could be wrong). b) arrange for an exemption for any GPL software from those patents. c) implement a codec with no known patent issues (like ogg vorbis). This is an issue which is very often overlooked in the hot GPL debates. However, in the commercial world, it is possibly the most important one. Not overlooked, it is just not an issue. Conclusion (IMHO of course): if you have the choice, use a license that is OSI-compliant but does not have the anti-patent clause. Or has it phrased differently. It all depends on what your goal is. Remember that the GPL also offers protection to companies. One of the reasons companies like IBM and SGI are releasing some of their stuff under the GPL is precisely because it does protect them from having their competitors simply take the technology and incorporating it into their non-open source software. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
