Re: [asterisk-users] Trigger Asterisk after data inserted in mysql
It looks like the answer is yes. http://crazytechthoughts.blogspot.ca/2011/12/call-external-program-from-mysql.html From the page, here is code to execute a UDF library and call a shell. Clearly there would be a heavy penalty to launching a shell so you would want to carefully evaluate the frequency this is executed on your system. | DELIMITER @@| |CREATE| |TRIGGER| |Test_Trigger | |AFTER| |INSERT| |ON| |MyTable | |FOR| |EACH ROW | |BEGIN| |||DECLARE| |cmd ||CHAR||(255);| |||DECLARE| |result ||int||(10);| |||SET| |cmd=CONCAT(||'sudo /home/sarbac/hello_world '||,||'Sarbajit'||);| |||SET| |result = sys_exec(cmd);| |END||;| |@@| |DELIMITER ; | -dbc Message: 1 Date: Tue, 18 Sep 2012 15:41:46 -0400 From: Ahmed Munir ahmedmunir...@gmail.com Subject: [asterisk-users] Trigger Asterisk after data inserted in mysql To: asterisk-users@lists.digium.com Message-ID: CAGMN=JdbE5FdDSQXxZ9OrWXu3Pvgc-hj-EnPxUrG=rjhgsd...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi all, I would like to know, is there a way to trigger Asterisk after data inserted into mysql DB? Like here what I'm trying to do, when the new data inserted into MySQL DB, it sends the request to Asterisk along with the new data (that is inserted in DB) for making outbound call i.e. Realtime. Currently I've set a cron job that execute my script every 30 seconds and checks for a new data in DB. If new data is inserted in 30 seconds that script will run and sends the data to Asterisk for making calls. (This is the case which I'm thinking to avoid) Please advise. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 63
I have 2 FXO channels from which I want to route incoming calls to different contexts in extensions.conf. I edited the context entries in dahdi-channels.conf and created matching entries in extensions.conf. One channel is routed to the new context as I want, but the other channel is stuck going to the default from-pstn context no matter what I do. Can anyone see what I've missed? From dahdi-channels.conf: ;;; line=3 WCTDM/4/2 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-3 channel = 3 callerid= group= context=default ;;; line=4 WCTDM/4/3 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-4 channel = 4 callerid= group= context=default You have multiple context= lines in your file and the order within the file is important. channel = should be the last item. So channel 4 is actually reading the context=default line which is 3 lines under channel=3 in your config file. -- _ -- 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] Interesting problem with IP's
Don't forget that many routers treat the designated private address space differently because it assumes the device is being implemented as a border router. In this configuration they block most traffic unless you specifically set rules to permit traffic to flow. -dbc. ___ -- 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] One side SIP goes dead on length conversation
Has anyone seen something like this before. Randomly, on longish calls, the local side of the call audio goes dead. Meaning remote caller can hear us but we cannot hear the remote person? Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. WANPIPE Release: 3.4.1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 169 | 4 | 1| N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT TE1 | N/A | Connected | Local sets are all Aastra 9143's. - dbc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OpenWRT
On Mon, 27 Jul 2009, Jeff LaCoursiere wrote: 1) The latest 8.09 kamikaze no longer supports the Broadcom radios, so ... Because of closed-source drivers the Broadcom chips only work on the 2.4 series kernels. OpenWRT does make a 2.4 kernel version _and_ a 2.6 kernel version. Use the 2.4 and the radios work fine. 2) I suppose this should have been clear to me from the start, but without an external (or hacked internal) storage of some kind, running asterisk on Make sure you have the right version number within the Linksys model. They changed drastically the RAM/Flash in the units (downward) as the production ran on. There are some charts online to go by. But the skinny is use a WRT54GS v4 or lower. V1.1 2 were the good ones with double the RAM. 3) OpenWRT seems to be less stable and not as mature as dd-wrt, which I I guess this is someone subjective and OpenWRT is somewhat in flux with 2 products under the same brand right now. White Russian was the previous release which is still available. Used predominantly NVRAM configs and had a smaller audience of platforms that it would support. It did however have a great GUI with lots of features. Kamikaze is the new version which has moved to more traditional config files and has an objective to be more platform agnostic. As a long-time White Russian user I admit the GUI has a long way to go before it can be considered a replacement for the White Russian version. I myself have never encountered stability problems with either version. Not sure how much DD-WRT has improved. A few years back OpenWRT was the clear winner (in my mind - no flames please) and I haven't re-evaluated the competition lately. -dbc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OpenWRT
Yeah, have it running on several units. It's really quite simple now. - Goto System - Packages - Scroll down to Update Package List and wait a few seconds for that puppy to refresh. - You now should have a list of installed packages followed by a very long list of available packages. - Find the asterisk version you want in the list and install it. - The asterisk package is just that, asterisk only. You will need asterisk-sounds for basic voicemail/ivr functions and you will also need the asterisk-voicemail package. Obviously either pick the 14 or the 16 tree as appropriate. -dbc. Date: Fri, 24 Jul 2009 12:12:35 +0200 From: abdelkader abdelkader2...@gmail.com Subject: [asterisk-users] Asterisk on OpenWRT Did anyone succeeded in installing Asterisk on OpenWRT system. pls help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OpenWRT
I have installed them on a Linksys WRT54GL or WRT54GS v4/v3/v2/v1.1 devices. My mother-in-law's runs fine and she doesn't notice the difference. I know that is very subjective but to be honest I never looked at it for more than home-use/1 line applications. Can't say I've had a problem that caused me to look at its load level transcoding. I can tell you she has been on the phone and received VM at the same time so there are two concurrent sessions. It means she keeps her number even though she moved to a retirement home that is out-of-area-code so she's more than happy. Plus calling between us is traditional 10-digit dialing although it is a SIP trunk - not that she (or my family) notice any difference. -dbc. From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Asterisk on OpenWRT What router did you install it on? Any stats on concurrent conversations / transcoding? On Fri, 24 Jul 2009, David Cook wrote: Yeah, have it running on several units. It's really quite simple now. - Goto System - Packages - Scroll down to Update Package List and wait a few seconds for that puppy to refresh. - You now should have a list of installed packages followed by a very long list of available packages. - Find the asterisk version you want in the list and install it. - The asterisk package is just that, asterisk only. You will need asterisk-sounds for basic voicemail/ivr functions and you will also need the asterisk-voicemail package. Obviously either pick the 14 or the 16 tree as appropriate. -dbc. Date: Fri, 24 Jul 2009 12:12:35 +0200 From: abdelkader abdelkader2...@gmail.com Subject: [asterisk-users] Asterisk on OpenWRT Did anyone succeeded in installing Asterisk on OpenWRT system. pls help. ___ -- 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] Need Recording Solution in Asterisk
One of our client Bank has 900 employees working in different locations. They need to record all internal and external calls. Can any body suggest Call Recording Solution for this requirement. We need to know the Hardware / Bandwidth and all requirements and costing. Few questions first 1. Why are they being recorded (business need)? 2. Does the value of the recording remain constant over time or diminish? 3. What criteria will you be required to retrieve the recording with? 4. Do you expect users to retrieve their own recordings or make requests of a records management operations staff? 5. Does everything need to be on-line or near-line/off-line and do you require a data management and migration solution? 6. Do you need to do word spotting and trend analysis on the content of these recordings (target marketing and customer service analysis typically)? Recording the call is quite easy. Storing it for retrieval which is acceptable to the business under their potentially diverse requirements is the tough part to nail down. There are commercial products like Witness out there which do a good job of this at a premium price. If the business drivers have low impact, you could simply record in asterisk and archive the files with some creative scripting and database work. You said this is a bank so I'm presuming they will have a formal risk analysis methods in place which would guide you through qualifying the requirements. Check out what the IT/CIO folks have to help you out in this manner. -dbc. ___ -- 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] Implementing an Asterisk Server behinda MeridianNorstar
Steve, Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1 card in the Norstar How does a user on the Norstar dial 221 and reach a voip only user connected to asterisk via ip only? That assumes as you mentioned new users are added as voip users in the future? Have the Norstar programmer send all 3 digit, unused extensions to the PRI. Then Asterisk will see 221, etc. and can handle at your dialplan sees fit. Retaining all NXX, NXXNXX, 1NXXNXX etc to the standard treatment they receive now. --- dbc. ___ -- 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] trixbox, sangoma a200, dell poweredge
Is there any reason why I should be experiencing such bad line quality on inbound calls from PSTN? Call quality is perfect when plugging in a regular analogue phone. Do you have other phone lines you can try the A200 with? Have you asked Sangoma support? Ditto on Sangoma support - they are excellent. Do you have hardware echo cancellation on this board? (Is there a D at the end of the model number?). Sangoma's hw echo cancellation is outstanding, if you are hearing training then I assume you aren't using it or the board doesn't have it. Assuming the PC is doing other only nominal things, yes, this PC is capable for what you are doing. (PC is NOT the VPN endpoint, don't have Tomcat, SQL or a spam filtering on a mailserver running on this box, etc.). Check that you are using IO-APIC and that everybody is getting their own interrupts. Do ifconfig w1g1. w1g1 Link encap:Point-to-Point Protocol UP POINTOPOINT RUNNING NOARP MTU:8 Metric:1 RX packets:1243390644 errors:0 dropped:0 overruns:0 frame:0 TX packets:1243390644 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:100 RX bytes:9947125152 (9.2 GiB) TX bytes:9947125152 (9.2 GiB) Interrupt:233 Memory:c206-61fff Check that errors/dropped/overruns are low if not 0. If they are not 0, re-run ifconfig and check that the numbers don't increase (they can have errors when the drivers first load and get synced up, but then stabalize. Be concerned if the number is not 0 and is over 1000. Based on your description of 2 fxo 1 fxs board, I think you actually have an A400 (as the A200 only accepts 2 modules total however it may still be reported as an A200 family - don't know, haven't used a 400 yet). Do wanrouter hwprobe to find out your info - note HWEC=32 means HW echo caneller, IRQ=233 (higher than 16) means you have IO-APIC activated. --- | Wanpipe Hardware Probe Info | --- 1 . AFT-A200-SH : SLOT=2 : BUS=5 : IRQ=233 : CPU=A : PORT=PRI : HWEC=32 : V=11 Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=1 A108=0 Armed with more info, Sangoma support (or us on the list) can help out more. dbc. ___ -- 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] question about queue
Two use-cases where autofill=no is desirable: 1) If it's important that you answer your callers in strict order (i.e. in order to meet estimated wait time commitments etc). Not always the case. Let's look at multiple queue assignment where agents have skills (logged in) to multiple queues. AGT1: Has SkillA, SkillB, SkillC AGT2: Has SKillA SLA: 24 seconds Senario Calls in Queues: Call1: SkillA - 15 seconds Call2: SkillB - 12 seconds AGT1 will become available in now() +2 seconds AGT2 will become available in now() +6 seconds CASE 1 (Calls in strict order): TIME=now()+2: AGT1 becomes available, CALL1 matched, time in Q now 17 seconds, assigned, SLA OK. TIME=now()+6: AGT2 becomes available, CALL2 NOT matched, not assigned, AGT2 idle, awaiting AGT1 to finish call, time in Q now 18 seconds. TIME=now()+10: AGT2 idle, CALL2 sitting in queue, SLA failed. CASE 2 (Calls not in order, system SMART enough to read into the queue and predict availability based on historical data) TIME=now()+2: AGT1 becomes available, CALL1 matches, but system knows that CALL2 is also a match and remaining agents are NOT a match. Predicted availability says call 2 will fail SLA, system assigns CALL2 to AGT2, time in Q now 14 seconds, SLA OK. TIME=now()+6: AGT2 becomes available, CALL1 matches and is assigned, time in Q now 21 seconds, SLA OK. TIME=now()+10: AGT1 on call 2, SLA OK. AGT2 on call 1, SLA OK. Now this isn't strictly the problem originally described but I'm trying to articulate where the use case as specified falls down in real-world environments. This also shows and area that Asterisk (and _many_ other switches) have not gone yet but we need to aspire to. This type of functionality is why you currently shell out the bucks for Avaya. - dbc. ___ -- 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] Microsoft Office Communications Server
Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? sarcasm What!? Microsoft implementing something not compliant with official standards. Your kidding? /sarcasm Sorry Matt, no advice here but I just couldn't resist. -- David Cook ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP Phone is really the best?
Seriously, if you intend on proposing this to a customer it means you are selling your professional services. If you are asking questions like this, how successful do you expect your customer engagement to be? Even if someone recommends the best phone for your particular application, you will still have zero competency with it and spend inordinate amounts of learning time and re-work on the customer's time. Your inexperience will show. Customers are demanding and you will get thrown out on your a**. People expect IT to fail from time to time (unfortunately), but they expect 100% availability from their phones. Anything less and you will find yourself with a priority meeting at the client that includes your manager, CEO and their lawyer. Nothing travels faster than a bad reputation. Walk away. Research. Build a lab. Learn. - dbc. From: William Herrera [EMAIL PROTECTED] Subject: To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com I need to quote a client for a job and I was just wondering. Out of all the IP Phones out there, which one is the best and why? Thank you all, all opinions will be accepted. William Herrera LAN/WAN Technical Consultant ___ --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] [on-asterisk] Configure one call per line on Cisco 7941/7961
Ahh. Differences with the 7961 software from that of the 7960's. Sorry, need to research more. - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-26-07 12:29 AM To: David Cook Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Re: [on-asterisk] Configure one call per line on Cisco 7941/7961 David, Yes, I'm aware of that, but unfortunately it does two calls on each line appearance (button), so the first two calls go on line 1, and the third will appear on line 2. I'd like to limit it to 1 call per line. Any ideas? Gary On 9/25/07, David Cook [EMAIL PROTECTED] wrote: Gary, if you register multiple lines with the same SIP credentials the phone will do rollover and take care of it. (2nd call comes in on L2, etc.) - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-25-07 6:37 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [on-asterisk] Configure one call per line on Cisco 7941/7961 Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961
Gary, if you register multiple lines with the same SIP credentials the phone will do rollover and take care of it. (2nd call comes in on L2, etc.) - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-25-07 6:37 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [on-asterisk] Configure one call per line on Cisco 7941/7961 Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
Date: Fri, 31 Aug 2007 13:19:32 +0300 From: Dovid B Subject: Re: [asterisk-users] phone as control interface (was 99 bottles of beer) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original ) Off-hook ) Dialtone ) Press ** (change to remote mode) ) To control the... ) Press 1 ) To change the vol... ) Press 1 ) To mut... ) Press 0 I am new to the whole controlling devices in your home from asterisk. Can anyone give me a URL to devices that I can connect to my box that can then connect to the lights, security system, TV etc ? This is a whole new area for me to play and get lots of sleepless nights ;) X10 control (Send data control signals over house wiring) I use an X10 Firecracker (CM17A) interface http://www.smarthome.com/1141.html which is a little radio transmitter the size of a DB9 shell and plugs into a serial port. The software that comes with it is for Windows and is very lame. However, there is a unix tool called bottlerocket which is a command line utility http://www.linuxha.com/bottlerocket/ to control the device. There are some smarter devices but that infers programming them within their constraints/user memory/etc. The command line one seems to work real well for me because the computer is far more capable than the other intelligent devices given the time to program it correctly. I have some code to calculate sunset so all my timings are relative to the correct sunset time so there is no altering for time of year or DST. This device can also send signals to more than one house code as I have two receivers. One for the lights stuff, and another for the sprinkler system. They don't make the one I have anymore, but here is a link to some others http://www.smarthomeusa.com/Shop/wgl-irrigation// X10 Warning: Read up on the technology. There are some controllers that are BI-DIRECTIONAL which means the receiving device will tell you what it did/what its status is rather than assuming it did what you asked it. X10 can have difficulty sending to some devices depending on which side (leg) of the power circuit you are on. (There are bridges to fix this problem too). X10 themselves also make some of the ugliest wall switches I have every seen. Leviton make x10 switches that are _really_ attractive (spouse friendly in your decor). They also work _much_ better with more consistent (virtually perfect) control. A much more professional system but be prepared to pay for the wife-approved model. Depending on features some of the Leviton versions are well over $100. X10 is also being replaced by a newer technology called Insteon. Don't have any of these devices yet but it looks like X10 version 2.0 and is backward compatible. Manual wired versions You can also get I/O interface boards for your PC which typically plug into a serial port and provide signalling to turn on/off many outputs with varying voltage/load characteristics like this http://www.sparkfun.com/commerce/product_info.php?products_id=20 PIC/Basic Stamp http://www.parallax.com/html_pages/products/basicstamps/basic_stamps.asp There are other intelligent devices like the PICs from Parallax called Basic Stamp modules. These are little computers designed specifically for I/O control type tasks. This is roughly the kind of little computers you might find in you microwave, etc. Only these ones are designed with an open-ended consumer programmable interface for creating general purpose devices. (These little guys also support a neat mode where you can create a master/slave network of many of them kinda like an RS485 industrial control bus. That means only one of these devices needs to connect to your PC but you could control hundreds of these in robotic control or data acquisition type scenarios. This is only the tip of the iceberg and I am certainly not the authority on this. But take a look at some of the links and let your imagination run wild. This is what got my daughter interested in programming. When she saw that you could get outside of the box and control real-world stuff from actions on the computer she was hooked. ___ --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] 99 bottles of beer
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Before this thread I already had a Firecracker on the server, a fair assortment of lights and the sprinklers are on an X10Pro Irrigation Controller. Damn, now I'm gonna be up all night. - dbc. ___ --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] Polycom IP 4000 Soundstation SIP Conference Phone Question
Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound? Works fine and sounds good. It's a Polycom so it has horrible webUI. You really should use config files for it instead. Remember with Polycom the field called network address is your _extension_ not your ip. - dbc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Integrated T1
On Thu, 24 May 2007, Jeremy Mann wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? The Zaptel/Asterisk infrastructure can definitely break particular timeslots out of the T1 for voice, but it is not my impression that any existing WAN drivers for Linux support Digium cards or cohabitation with Zapata and can give you a serial data interface on other channels. There are obvious risk factors with the scenario of your Asterisk box being your CSU/DSU/Firewall Router but for a small office this can actually be a good thing. Sangoma cards with their Wanpipe drivers can do this for you. dbc. -- David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk setup for church / conference call
Quoting Tim Litwiller to connect to the speaker system I either need to trigger a ring on a analog line to the phone interface on our speaker system, it picks up on the first ring, or we can manully push a button that picks up the line. If we do the second we would have to have something in asterisk connect it to the conference when it picks up. We just put a softphone on the PC that runs Easy Worship and plugged the soundcard output into a mixer channel and the input to an Aux out. Recording is fairly painless, just use mix monitor appl as part of the dial plan for the secret extension that launches the conference from the PC. This is assuming you have a manned sound reinforcement system. If your services are more low-key then the pastor will have to do it himself before the service starts. FYI: Churches are the _perfect_ example of a distributed business environment where Asterisk shines. What other company do you know of that has such a number of workers who don't have an office in the building? Our church is currently moving - possession June 30 - and we will have deskphones for offices, deskphones at home for key ministry leaders, softphones for minor ministry leaders and/or phantom VM's with email attachments. Don't for get aliases to job functions! People in ministry and volunteer can come/go. Make sure your IVR has name and functions. Somebody will call looking for Children's Ministry, or for the Pastor without knowing who that person would be. dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: New T1 Asterisk installation
Quoting [EMAIL PROTECTED]: I have two options, T1 or 15 analog lines. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf and zapata.conf 15 lines should be well beyond the cost justification point for a T1 and you will get significantly better quality (disconnects) and functionality out of digital trunk. Plus you clean up the telco closet. Remember, you don't need to activate all 23 lines so if you just need 15 then you can activate only that number. You also can have potentially hundreds of numbers that terminate on this group of lines. This makes some of your coding a little different because you no longer make an association between which port(s) ring and what number the caller dialed to get here. This is called DNIS (Dialed Number Identification System) (people don't flame me for the ANI/DNIS thing OK? Not relevant for this discussion). When ordering the PRI the telco will ask you what type of signaling you want and how many DNIS digits. Personally, as we have intermixed area codes, I always ask for 10 digits DNIS. This means when asterisk answers the phone the $EXTEN will equal the full phone number the caller dialed to get here. loadzone= us defaultzone = us span= 1,2,0,esf,b8zs bchan = 01-23 dchan = 24 span= 2,3,0,esf,b8zs bchan = 25-30 dchan = 48 This is a zaptel.conf for 2 PRI's. 23 chan on 1 and 6 on the second. It stipulates ESF (Extended SuperFrame) with b8zs coding. Both PRI's have their D channel on the last (24th) channel. As for emulation I try to ask for NI2 (which is a config that goes in zapata.conf for switchtype). Hope this gets you started. -- dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Which SIP phones to buy?
Quoting Stephen Bosch [EMAIL PROTECTED]: I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge. Advice, anyone? -Stephen- You said 'office' so I'm presuming you want business quality. If you have already tried the Polycom's I'd look at Aastra (just did a 50+ seat implementation with 9133i's 480i's) and also look at the Cisco 79xx's. Cisco's Aastra's both handle multiple appearances differently but both are excellent. Cisco has superb handsfree quality. Aastra has better BLF support. You will have to evaluate for yourself. Aastra is significantly cheaper. That said, there is a 7960 on my desk that isn't going anywhere soon. I hear the Grandstream firmware is better now but physically they are still pretty flimsy. I would stay away from them for anything but experimentation. dbc. ___ --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] Cisco IP Phone services.xml sample?
Does anyone have a small, plain services.xml file for a cisco ip phone, preferably one that will work on a 7960? I can't seem to get my xml right, and no matter what I send to the phone I keep getting parse errors. Thanks Shawn CiscoIPPhoneMenu TitleXML Portal/Title PromptChoose from a range of XML Services:/Prompt MenuItem NameBerbee XML Main Menu/Name URLhttp://phone-xml.berbee.com/menu.xml/URL /MenuItem MenuItem NameBT Exact XML Main Menu/Name URLhttp://193.113.58.136/bt//URL /MenuItem MenuItem NameStock Quotes/Name URLhttp://phone-xml.berbee.com/cgi-bin/stockchk.pl/URL /MenuItem MenuItem NameUS Weather/Name URLhttp://phone-xml.berbee.com/cgi-bin/weather.pl/URL /MenuItem MenuItem NameUK Weather/Name URLhttp://193.113.58.136/bt/weather/weatherinfo.asp/URL /MenuItem MenuItem NamePhil's XML Development Page/Name URLhttp://flame.tiefighter.org/fwd/xml/dev//URL /MenuItem /CiscoIPPhoneMenu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: how to define a pilot number
is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Lito I'm assuming you are talking about analog lines as PRI's will do this more-or-less naturally. This is a telco feature as opposed to an Asterisk feature. Here in Bell Canada country they call it Ringer Equivalence. Call your local carrier and they should be able to tell you what they call it in their marketing world. You tell the telco which lines you want the calls to roll to then all three will terminate calls to the pilot number. Now it doesn't work exactly as you had described - it doesn't move the call so as to free up the first port. It merely says the first port is busy and terminates the next call on the next port in sequence. This means you can't count on which line is available at any time. For outbound, you need to put the three lines in an Asterisk group and test the group for availability to select an available line to dial out on. dbc. -- David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106
Lito Lampitoc wrote: thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? I don't think it's random. I think its just sequential. If main line is busy, try second. If that is unavailable, then try third in sequence, etc. It's called rollover here. Correct, its in the sequence you told the carrier you want. Caveat, You _can_ have contention with analog lines. Meaning someone calling in at precisely the same time as someone calling out - not often, but it will happen. To help aleviate this, get the carrier to roll the lines 1-2-3 and outbound you pick the lines 3-2-1. - dbc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 48
From: Ricardo Carvalho [EMAIL PROTECTED]
Subject: [asterisk-users] How to match wild card inside a GoToIf?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3)
exten = s,2,Hangup
You are going to need a substring of the original. I'm thinking
something like the following although I haven't tested it.
exten = s,1,GotoIf($[${ARG1:3} = 220408]?2:3)
dbc.
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[asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
to do it is archaic. What?!?! The Dell tech guy kept saying that I can define an IRQ in Linux, and I kept telling him that I need two unique (not Doesn't IO-APIC work for you or is that what you meant by virtual IRQ? I thought IO-APIC changed the way the APIC worked but it was under OS control and therefore they could put smaller/simpler/cheaper BIOS in the raw box. Please correct me if I'm missing the boat. (I had a sharing problem in my PowerEdge 1400SC and IO-APIC seemed to fix it up nicely. The server has been in operation for 6 yrs now with Asterisk running on it for the past 3). David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Dell Server Question
Quoting Nick Whitaker [EMAIL PROTECTED]: The problem I'm having is the only PCI slot shares an IRQ with the SATA controller. Any altering of one device's IRQ takes the other device's IRQ with it in lockstep. Nick, the word from Dell is that SC stands for Simplified Configuration and there is less ability to move stuff around as you wish. I too have a PowerEdge SC series (SC1400) which caused me some of the same grief you are experiencing. My basic understanding is that some of the PCI IRQ's are tied together as there is less hardware/firmware support and is one reason the units are so price competitive. Don't get me wrong. I love the box for it's price/performance point and it has been rock solid for 5 yrs. I fixed this by changing the linux kernel to include IO-APIC support which permits the OS to route interrupts without overlapping IRQ's. I'm assuming any reasonably new Dell hardware will support this and it comes on by default in most SMP distributions. You then get IRQ's ranging into the hundreds with no overlap. Note the eth0, Cyclom-Y, 2 SCSI's Sangoma which used to share in the old scheme getting IRQ's into 3 digits. # cat /proc/interrupts CPU0 0: 2850398012IO-APIC-edge timer 1:952IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 11: 0 IO-APIC-level ohci_hcd 12: 3894IO-APIC-edge i8042 14: 64252737IO-APIC-edge ide0 177: 52753938 IO-APIC-level eth0 185: 260531 IO-APIC-level Cyclom-Y 193: 25788929 IO-APIC-level aic7xxx 201: 30 IO-APIC-level aic7xxx 209: 2849304364 IO-APIC-level wanpipe1 NMI: 0 LOC: 2850775576 ERR: 0 MIS: 0 dbc. -- David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Running Asterisk on a Home rotuer
On 12/7/06, Dovid B [EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid Sure. I have 5 units out there on Linksys WRT54GS v1.1 through v4 units. The software is OpenWRT.org. Asterisk is simply an available package to load once you have replace the original firmware with OpenWRT. There are several models that can run the software. Check the HW compat list on the site. They go right down to revision numbers identified by serial # patterns. Be careful of the amount of RAM they have. You will be storing voicemail in RAM unless you put it off-device like an NFS mount, etc. (Some mfg/models have USB2 ports and you can put a USB stick on them and basically forget about the problem). -- David Cook (Canada) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
Vincent Delporte wrote:
Hi
Most of our customers have generic names like Hospital, so I need
to
rewrite their caller ID name by looking up the number in a database
on
the Asterisk server, and rewriting the name such as Reading
Hospital
so that we know who's calling.
Any idea if this can be done with Asterisk, and how to do it?
Little C and AGI to the rescue (uses MySQL too). DB schema in the code
comments at the top.
dbc.
extensions.conf:
;; Advantech primary context (Sangoma A200D ports 12);;;
;; Primary telco number (905-xxx-)
;
[advan-primary]
exten = s,1,NoOp(Primary line - ${CALLERID}) ; write log entry
exten = s,n,agi,clid_override|${CALLERID(NUM)} ; CLID agi override
exten = s,n,Goto(cook-main-menu,s,1) ; Jump to main menu
exten = s,n,Hangup ; end/fallthrough
clid_override.c:
/* clid_override.c
* (c) Advantech Systems Integration, 2006
* Author: David B. Cook, [EMAIL PROTECTED], 905/xxx-
* Initial Delivery: Version 1.0, March 1, 2006
*
* Application to set the CLID NAME field from a local database
* when the field comes in empty from the carrier.
*
* Meant to be called from Asterisk as an AGI lookup
* Connects to MySQL database : CLID_NAME
*
* Database definition
* # Host: localhost
* # Database: asterisk
* # Table: 'CLID_NAME'
* #
* CREATE TABLE `CLID_NAME` (
* `CLID_NUM` varchar,
* `CLID_NAME` varchar,
* PRIMARY KEY (`CLID_NUM`)
* ) TYPE=InnoDB; * CLID_NAME
*
* Modification History:
* XXX 00,00 dbc - Example modification history format
*/
#include stdio.h
#include stdlib.h
#include mysql/mysql.h
#include string.h
#if !defined(MYSQL_VERSION_ID)||MYSQL_VERSION_ID32224
#define mysql_field_count mysql_num_fields
#endif
#define SELECT1_QUERY select CLID_NAME from CLID_NAME where
CLID_NUM='%s'
int main(int argc, char **argv)
{
MYSQL mysql,*sock;
MYSQL_RES *res;
MYSQL_ROW row;
char *DBhost=put hostname here;
char *DBuser=put MySQL username here;
char *DBpw=put MySQL password here;
char *DBdb=put MySQL database name here;
char qbuf[512];
int i=0;
char line[80];
/* use line buffering */
setlinebuf(stdout);
setlinebuf(stderr);
/* read and ignore AGI environment */
while (1) {
fgets(line,80,stdin);
if (strlen(line) = 1) break;
}
sprintf(qbuf,SELECT1_QUERY, argv[1]);
/* debug: show query formulation */
/* printf(SQL: %s\n, qbuf); */
/* Initialize and connect to the server */
mysql_init(mysql);
if (!(sock =
mysql_real_connect(mysql,DBhost,DBuser,DBpw,DBdb,0,NULL,0)))
{
fprintf(stderr,Couldn't connect to
engine!\n%s\n\n,mysql_error(mysql));
perror();
exit(1);
}
/* Perform query to determine if a row exists in the database for the
* CLID in question.
*/
if(mysql_query(sock,qbuf))
{
fprintf(stderr,Query 1 failed (%s)\n,mysql_error(sock));
exit(1);
}
/* No results - fatal error */
if (!(res=mysql_store_result(sock)))
{
fprintf(stderr,Couldn't get result from query failed\n,
mysql_error(sock));
exit(1);
}
if(mysql_num_rows(res)=1) {
/* CLID is PK so should only be 1 row, but I'm going to*/
/* say = just so it won't break if no PK and multiple hits. */
/* If so, will just re-set CLID again but won't break Asterisk */
while(row=mysql_fetch_row(res)) {
printf( Set VARIABLE CALLERID(name) \%s\ \n, row[0]);
/* send the output back to Asterisk */
fgets(line,80,stdin);
fputs(line,stderr);
}
}
/* Clean up memory tables/free resources */
mysql_free_result(res);
/* Terminate the database connection */
mysql_close(sock);
exit(0);
return 0; /* Keep some compilers happy */
}
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[asterisk-users] Re: Queues and multiple lines
Michael Sampson wrote .. Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An agent can then put their current call on hold and go to the new call, say something like thanks for calling please hold, then go back to their first call, finish it up and then go back to the second call. Michael, I don't think you want to do this in a Contact Centre environment. Remember that once the agent has answered the call you have now locked the caller to that agent. If another agent becomes available first, they will no longer get the call. The free agent will sit idle (or get the next call in queue which is NOT the caller who was answered). The caller who was answered on line x by the other agent must wait in perpetuity for the agent to become available, yet their TALK TIME clock is running as the call WAS ANSWERED and ASSIGNED to the agent. You are better to play announcements during the queue wait time to say whatever you want communicated to the people in queue. This way they maintain their position in queue, the availability to be assigned to any available agent that becomes available and their call stats work out. The call stats are really important as this is how you are going to measure you agents. Even if you can separate the hold/talk times, your stats for the agents will become meaningless and hurt your Work Force Management (WFM) programs and seriously impair you ability to manage/measure your people. dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Polycom provisioning and Pure-FTP : problems
Mike wrote: PS: If there is a better FTP server suggestion Ill take it, but one of my must-haves is easy of use and virtual users functionality (with different chroot folders). I don't know whether it supports the specific functionality you require, but we have always uses vsftpd with no problems. Best regards David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7960 - Fast dial
Yes it can be configured on the phone. Settings - Call Pref - Speed Dial Lines - pick your button edit Putting something here overrides the ability to use it as a line button and changes the icon to a dial pad. (I have programmed my MoH extension on Line 6 so I can listen to my mp3 catalog) dbc. From: Tomislav Par?ina [EMAIL PROTECTED] Subject: [asterisk-users] Cisco 7960 - Fast dial Cisco 7960 has six buttons/lines. Can some of them be configured for fast dialing? If it can't be configured on the phone, how can I configure it on Asterisk? ___ --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] 7960/SIP MWI Question
The 7960's have an envelope that appears in the display next to a line which has voicemail. Also, the MWI light is a logical OR of all the defined lines. Is there a way to tell the phone NOT to display the MWI for certain lines but retain the envelope for all? If you get enough VM on busy lines then the light tends to lose meaning and you may as well have it on all the time! I'm currently on POS3-06-3-00 dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk to do remote control
If you just want to control a couple of digital points this hardware may be overkill, but it is cool stuff. For smaller implementations you can just use the outbound control lines (DTR RTS) on an RS232C port. That can give you control of two on/off devices. They only sink about 20ma so isolate them with a solid state relay or something. A C program to turn on/off is fairly trivial and run it from AGI. I don't want to clutter the list with code but I can supply if anyone needs it. dbc. -- David Cook ___ --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] FOP v.27 IAX trunks not ringing
I am using FOP .27 and I have Zap IAX trunks. Although the IAX trunks do show and appear registered (not dimmed) on the display, they show no activity while in use. Any ideas?? Segments of op_buttons.cfg iax.conf are included: op_buttons.cfg [Zap/1] Position=23 Label=Cook (Main)%0a(905) xxx- Extension=-1 Icon=0 [IAX2/416xxx] Position=24-26 Label=Personal Line%0a(416) xxx- Extension=-1 Icon=0 [IAX2/647yyy] Position=27-28 Label=Business Line%0a(647) yyy- Extension=-1 Icon=0 iax.conf ; Registrations for remote IAX servers (dynamic config) register = 416xxx:[EMAIL PROTECTED] ; Personal register = 647yyy:[EMAIL PROTECTED] ; Business [416xxx] ; Unlimitel DID - Personal username=416xxx type=user context=DID-incoming [647yyy] ; Unlimitel DID - Business username=647yyy type=user context=DID-incoming Thanks, dbc. ___ --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] Dialplan or matching
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those Doh! moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3=800, etc. rather than by 2's, (so it doesn't overlap with
8XX area codes) and select my local loop as the first pick.
dbc.
Kevin Smith wrote:
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2
= 66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 = 88)?TRUE:FALSE
exten = _18XXNXX,n(TRUE),Dial()
exten = _18XXNXX,n(FALSE), HangUp()
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[asterisk-users] Dialplan or matching
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort of like the SPA's can? Tollfree numbers for example. I can have a line for each combination: exten = _1800NXX, Dial, exten = _1866NXX, Dial, exten = _1877NXX, Dial, exten = _1888NXX, Dial, But I want to do is something like this: exten = _18[0678][0678]NXX, Dial, . Or to prevent the logic error which albeit small, the above would create: exten = _18[00,66,77,88:2]NXX, Dial, .. (representing that the next 2 chars must equal any of '00'.'66','77' or '88' Is there any syntax that allows this?? dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Sipura SPA-3000 vs Sangoma A200
I echo (pun intended) Rich's response. The Spa3k is ~ok~ but echo has always been a problem for my home office. The A200D works flawlessly. I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA motherboard as an always on, low powered solution. I have seen an A200D in a soekris 4801 (http://www.soekris.com) box running astlinux. I say saw, because it was at a show and the box wasn't plugged in. It was Jim VanMeggelen - one of the authors of the O'Reilly Asterisk book. You might want to drop him a line. The Sangoma has a 4-pin molex for power supply connection to augment the PCI bus when you need to generate ring voltage for FXS ports. The soekris (by default) won't give you that so either you put FXS external or you figure out how to get +5/+12 VDC to the Sangoma. Actually, you may want to check with Sangoma ... maybe you only need 5 or 12 but they just match the molex to be compliant with all PC hardware. I am trying to find out the differences between a solution using an external ATA (like the Sipura SPA-3000) or an internal PCI card (like the Sangoma A200 with 2 FXO 2 FXS ports). The nice thing about the SPA3K is that upon registration failure or power failure the FXO FXS ports get hardwired together so you get a power safe environment. The nice thing about the Sangoma is that it supports ring contexts by distinctive ring. I believe this is also called Ident-a-call in many places. For a home office this is great. I have a second number that rings my primary line with a different ring pattern for ~ 4.00/mth. rather than the expense of a second line. I program that ring pattern into zapata.conf and push those calls directly to Zap/4 (my fax) and other calls to Zap/3 (my house), etc dbc. ___ --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] Two phone numbers, one SIP provider
From: Filip Dr?gowski [EMAIL PROTECTED] Subject: Re: [asterisk-users] Two phone numbers, one SIP provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-2; format=flowed I think that context=incoming-[number] in firends definion is used rather for determinig context for outgoing calls. In sip.conf [general] section there is context= and register=/[extension] i think that extension in register line should be in context specified in [general] context=[context] register = :[EMAIL PROTECTED]/ register = :[EMAIL PROTECTED]/ I am only using inbound IAX but I do have multiple DID's from the same provider and I hope the action of parsing the files is the same. I don't think you want the extensions listed after the host. The following is what I have for my IAX trunks ; Registrations for remote IAX servers (dynamic config) register = 416xxx:[EMAIL PROTECTED] ; Home line register = 416yyy:[EMAIL PROTECTED] ; Business line Both calls happily go to contexts [416xxx] and [416] respectively. dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New Digium Card b410p
Tommaso Calosi wrote: Who knows something interesting about the new BRI digium card b410p ? For example, will it use the misdn driver or the native zaptel? Any interesting links will be appreciated too. ___ --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 The new 8 channel BRI card uses mISDN. According to Digium the hardware is finalised and they are currently beta testing the drivers. I was talking to Matt, one of the Digium developers that has been working on the card, so this is all first hand information rather than rumour or hear-say. Should be available worldwide through Digium's normal distribution channels in the next few weeks. Like buses (so we say in the UK), decent BRI hardware comes all at once. Xorcom are just about to release BRI versions of their Asterisk specific channel banks as well. Best regards. David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: MWI on Treo 600/650
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: MWI on Treo 600/650
I've been working on this off and on for AGES. There are some SMS portal sites that claim to be able to do this as well, but I have not managed to find one. I had found a company called bahamasystems which has an asterisk interface but it's a service and it's expensive. Another poster pointed me at nowsms.com. Looks a little more attractive (except for the Windows gateway part). However, I have not been able to find out the actual codes. Just the 111# stuff to get a return receipt, etc. ___ --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] zapata configuration parsing
Hi gang. Just put an FXS port on a Zap interface for the first time. I can't figure out which parameters in zapata.conf are global and which ones can be channel specific nested. I have mucked around with it but I can't seem to make any effect on the gain levels on a per channel basis. dring1context=pbx } dring1=0,0,0} obviously global because it sets conditions for dring2context=fax } all inbound calls dring2=387,321,0} signalling=fxs_ks } is this the lead or should channel be the lead group=1 channel=1-2 rxgain=6} can this go here to effect just chan 1-2? txgain=0} signalling=fxo_ks group=2 mailbox=500 channel=3 rxgain=0 txgain=0 mailbox= channel=4 rxgain=0 txgain=0 Thanks, dbc. -- David Cook (Canada) ___ --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] Polling Asterisk for Life
Obviously if Asterisk keeps going down there is another problem to be found. However, why not start it from /etc/inittab with respawn??? Else, poll from cron or a script with ps ax | grep asterisk | grep -v grep | wc -l to find out if it is running. dbc. Date: Thu, 2 Mar 2006 22:01:01 +0200 From: Cosmin Prund [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polling Asterisk for Life To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii AFAIK there are problems with repeatedly connecting and disconnecting the manager interface. Also you're probably using a proxy (all manager interfaces I've seen are using proxies), it might not be a good idea to pool something out of the manager that often. Did you consider running a cron job on the server, using asterisk -rx to run a command and then decide rather asterisk is down or not based on the result? This way you'd be doing on the server, working around the problems with the manager interface and saving some bandwidth :) . You might also be able to call /sbin/reboot directly from the cron script! If on the other hand the whole server is going down you may simply use ping! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, March 02, 2006 7:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polling Asterisk for Life Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking I could poll my SIP account for the CallWaiting value, but would like something that was not linked to my account. Just something that returns a single line is fine. ___ --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] UK ISDN2e with DDI?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dogers Sent: 07 December 2005 16:24 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK ISDN2e with DDI? Quoting John Daragon [EMAIL PROTECTED]: Patrick Lidstone (Personal E-mail) wrote: We're about ready to go ahead with a nice 6 line (maybe later 8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card. Before we do, could anyone confirm for me that BT's ISDN2e lines do actually provide Asterisk with the DDI number? We need to be able to route incoming calls based on which number is ringing. Yes they do. For DDI ranges you'll need to ask BT for a System Access installation (sometimes known as Point-to-Point) and configure the Junghanns appropriately. I'll have to check that, I guess - find out what they're set as! I'm probably just an old fogey with a programming background, but I find straight Asterisk *so* much easier to configure than [EMAIL PROTECTED] True, I've used bare Asterisk at home for my small get up, but [EMAIL PROTECTED] just does everything we need it to do here at the office (including the nice and pretty call log side of things that AMP provides!) When you say ringtones, do you mean sounds like a UK phone when it rings, or sounds like a UK phone when we ring someone else ? It does actually sound okay when we ring someone else, but when it rings, it has the long single american style ring. I've come across a few places that claim its built into the Grandstream and I'd have to create and upload a new one.. but I've also found others that say to edit various config files, which has had no effect (indications.conf and zaptel.conf both have the zone as uk.. Theres nowhere else it needs to be set, is there?). Andrew Try adding the following to your handset config in sip.conf. This forces the SIP device to get it's ring tones from Asterisk. Worked for us in v1.0.9 with Polycom handsets. progressinband=yes Be careful when ordering an ISDN2e line from BT. By default they come configured as Point-to-Multipoint with any additional numbers as MSNs. Most PBXs are better with ISDN2e Point-to-Point with DDIs, but BT then sting you for a £100 DDI planning fee in addition to the ISDN2e installation. One thing to consider is that DDIs are allocated in contiguous blocks of 10 numbers e.g. 0115 7889100 - 7889109. MSNs however are purposely allocated by BT randomly in what ever quantity you require. Officially you cannot have contiguous MSNs which aren't so good for PBX use. If you want inbound CLI display (CLIP) and/or the ability to specify the outbound number you are presenting as a CLI (CLOP/COLP depending on who you are talking to) this needs to be specified as well. By default you get neither but both are non-charegable upgrades (in our limited experience). David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: 4 HFC cards
Colin, I don't think that I'm the best person to answer this. I have an HFC ISDN card back in its box at work which I failed to get a decent connection when using it in combination with a Dell Optiplex GX240, Centos 3.4 and brstuffed Asterisk 1.0.9. The channels came up and calls would connect in and out, ableit unreliably, but the audio quality was very choppy. I suspect this was due to IRQ conflicts and it was impossible to assign a unique IRQ to the HFC card. As this was for a pilot PBX at a remote site, we went for a Multitech ISDN/VOIP Gateway. That Asterisk server just handles SIP IAX traffic now. We have a couple of Supermicro 5014C-MFs which I will go back to with the HFC card http://www.supermicro.com/products/system/1U/5014/SYS-5014C-MF.cfm though ultimately we will need to terminate an ISDN30e connection for our main site. I'll copy this to the list where somebody else may be able to help out. I understand the Florz patch may help you (http://zaphfc.florz.dyndns.org/) and this post to the list may give you a start on how to best handle multiple HFC cards in the same box: http://www.voip-info.org/wiki-Asterisk+zaphfc+install26 Regards David Cook From: Colin Whittingham [mailto:[EMAIL PROTECTED] Sent: 10 November 2005 05:00 To: David Cook Subject: 4 HFC cards Hi Dave, I have a site running with 4 hfc cards installed. This is running on an AMD Processor 2800 with 512 MB ram on a standard motherboard. If there were 5 PCI slots on the motherboard I am sure that it would work too. I do however have a question. I have recently tried to install HFC cards on AAH 1.5 using bristuff-0.2.0-RC8o. The cards initialise but I cannot make calls, the B channels do not seem to come up. Any ideas? Regards Colin ___ --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] Multiple zaphfc cards (for ISDN BRI) in a single machinemachine
Me thinks it is time for ISDN30e and a TE110P ;-). David Cook -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: 03 November 2005 10:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Multiple zaphfc cards (for ISDN BRI) in a single machinemachine Hello all, I've just returned from a visit to a client site where their existing incoming lines are in the form of 5 ISDN BRI connections (for 10 channels total). We have successfully deployed Asterisk boxes with 2 HFC-based cards in the past, but I've no idea how well a standard PC will handle 5 or 6 cards - i.e. every PCI slot has a BRI card in it. Any thoughts from folks who've tried this in the past? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (01604) 808408 Mobile: (07811) 332969 Skype: minotaur-uk ICQ: 13350579 AIM: MinotaurUK MSN: [EMAIL PROTECTED] Y!: Minotaur_Chris This email is made from 100% recycled electrons ___ --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 ___ --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] GSM cards / mobile phone cards for Asterisk?
If anyone knows of smaller-scale units that work on GSM900 and 1800, I'd also love to hear about them. You might want to investigate a Nokia 22 (http://europe.nokia.com/nokia/0,8764,56024,00.html). This provides a single GSM line which is interfaced to the PBX by an anlogue trunk/extension. From memory they cost around £100-150. I am going to revisit this as a solution to our ever increasing PSTN-GSM call spend as soon as we have our Asterisk PBX in place. David Cook JP Computer Services Delivering Business Benefit http://www.jpcompserv.co.uk ___ --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] Ignore incomingcall?
Use a separate context for each Dring.
dring2 cadence 0,0,0 will identify the primary number not the secondary.
If you want dring1=main number dring2=distinctive ring num then you
need dring1 as 0,0,0 and dring2 as the alternate cadence.
This context will ignore the calls on the main number if dring1context
is set to primary in zapata.conf.
[primary]
exten = s,1,NoOp(${CALLERID})
exten = s,2,Hangup
Is there a way to tell asterisk to ignore an incoming call?
I am using distinctinveringdetection and I am only interested in
answering
calls
on the 2nd number. Any call to the main line should just be ignored.
right now I have a context set for dring2 cadence 0,0,0
exten = s, 1, wait(30
exten = s, 2, Hangup
--
David Cook
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[Asterisk-Users] IAX register confusion
I have been unable to understand the connection between an IAX
registration for dynamic IP assignment and and the host definition.
I have signed up with an ITSP for a DID. My ip is dynamic and although I
have a dynamic DNS name, we are registering and outbound works fine.
I'm at a loss to understand the relationship between the registration
and the [section] definition in iax.conf that will allow me to specify
my context for inbound calls and deal with the inbound DID.
For example:
register = myuser:[EMAIL PROTECTED]
;OK. This part works fine. My dial statement calls
; exten =
_NXXNXX,3,Dial,IAX2/myuser:[EMAIL PROTECTED]/${EXTEN},45,tr)
;
; VoIP Local service from myitsp
;[something] ???
[LO_TRNK_MYSWITCH]
type=peer
host=dynamic
context=from-myitsp
secret=mypasswd
qualify=3000
; How do I construct this entry? I would _like_ the entry to be labelled
; LO_TRNK_MYSWITCH so I can maintain a naming convention that makes
; sense.
; How do I associate this with the inbound itsp so the calls come into
; the s extension in a particular context so I can deal with the DID?
I simply don't see how I associate the inbound stream with my section
heading?
Thanks, dbc.
--
David Cook
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[Asterisk-Users] Re: Asterisk and Dell SC420 Server
Just so you know, Digium doesn't recommend using this server (or a number of others from Dell). I think this is mainly because you can't choose IRQ's in the BIOS, and your Digium card may end up sharing an IRQ with an onboard device like the ethernet interface or the disk controller. To get around this, you can usually switch your Digium card to another open PCI slot. There may be other reasons that Digium doesn't recommend using Dell servers. You might want to ask them. - Noah I discovered that SC in Dell parliance means Simplified Configuration. Or in other words, limited control of IRQ's. There is also some meaning to the 2nd digit that inferred PCI2.1 and PCI2.2 but I don't recall what the formula is. Maybe someone else knows. For the record, I have an 1400SC which performs flawlessly for everything _except_ asterisk as I can't get my X100P not to share. For general purpose computing I am a huge Dell fan on quality, performance and price point but this disappoints me. -- David Cook ___ 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] How do I know that my machine will support APIC?
Regarding the SMP and interrupt issues. I know my machine is not running APIC now, but how do I determine if it is capable? Can I find out from the running system or is this something I need to know from the mfg? Currently the X100P shares IRQ with the secondary SCSI (yeah, go ahead and laugh). The box is a Dell PowerEdge 1400SC. Apparently the SC means Simplified Configuration and limits options on IRQ's among other things. dbc. -- David Cook ___ 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] RE: Connecting 2 * Together-Pulling hair out
http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers I have posted a doc on this to the wiki. Fist time poster. I couldn't figure out how to escape square brackets and tables looked like I would be there all day. Be nice :-) dbc. David Cook ___ 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] Re: Asterisk-Users Digest, Vol 8, Issue 119
Quoting [EMAIL PROTECTED]: Date: Mon, 14 Mar 2005 22:23:54 -0700 (MST) From: Greg Hill [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How NuFone.Net's customer service works. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII I'm always surprised by how many people claim to use NuFone.. I've tried, on more than one occasion, to contact them both by phone and email. After waiting on hold for a while, their phone system offered to let me leave a message and somebody would contact me as soon as possible. I did so, but never heard from them. Not in an hour, not the next day, not even within a week. Never. Peculiar sales strategy, to say the least. Maybe I'm lucky. I decided to give them a whirl last year. I thought their website was pretty weak (but then again, so's mine). I did have a problem signing up, some problem with the registration server. However, I sent an email to support then I took my family out for dinner. Upon returning home I had a response complete with confirmation, registration and config in my mailbox - and if memory serves, this was on a weekend. I immediately set it up and have been using it since. I'd like to see an export of the CDR so I can do something with it down the road if I wish, but all-in-all, I'm quite happy. dbc. -- David Cook ___ 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] A bit of a survey: What do do if you need more than 4 C.O. lines
I corresponded with Voicetronix around Christmas last year. Jim, there is a dealer in Ottawa although I got better answers from emails to Aus. There are two things that they don't do that the Zap cards do: Distinctive Ring Detection and fax detection. They went out of their way to say they were customer driven and features get in because customers ask. The gentleman made a claim of effort to get fax detection to work which sounded like it was a no-brainer in their code. If it is easy as claimed, I would expect to see it appear just because I enquired. I am particularly interested in the Dist Ring Detection however for they make cheap DID's for low volume like home offices, dedicated voicemail numbers, etc. David Cook From: Jim Van Meggelen [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4C.O. lines To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1250 I haven't followed this thread closely but have you looked into the Voicetronix OpenSwitch cards? http://www.voicetronix.com.au/hda.htm I've looked at them, but never heard much about them. Is anyone using them? Can anyone give a comparison vs. the TDM400? I'm using a Voicetronix OpenLine4, and it works well under asterisk. Initially I had some echo problems, but Voicetronix support is excellent and solved them (I've just updated the wiki with the bal# values they gave me). I can't compare it to the TDM400, not having used one, but you can use multiple Voicetronix OpenSwitch 6 and 12 cards in one system without the interrupt problem of the TDM400. That sounds like the ticket, then. ___ 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] Re: Asterisk-Users Digest, Vol 7, Issue 260
From: James Bean [EMAIL PROTECTED]
Has anyone every setup an external open/close relay, off say a serial
interface, and have an extension trigger the relay?
The following will do the trick. Just add a 5vdc solid state relay
('cause you can't sink too much current out of the RS232C port).
Substitute 2, 4 or 6 in the code below to turn on either DTR, RTS
or both signals. 0 for off.
Change SWDEV in the lpswitch.h file to be the serial port you intend to
use for the relay. I'm using some optically isolated relays I found in
town for $5.00 Cdn. The box to put it in cost more than the relay.
There is a bunch of extra defines in the .h file that were needed for
the larger project this was part of. Just ignore them, they won't hurt.
Call this program from your dialplan, and voila.
Compile with cc -i lpon.c -o lpon
/*
* lpon.c Lineprinter ON
* *** test program only **
*
* (c) David Cook, 1994
*
* Set signlal lines on serial port to turn on 5vdc
* signal. Used for solid-state relay (low current
* draw on RS232C port) to switch high voltage/high
* current load for printer.
*
* Part of an intelligent print spooler to only power
* on/off high draw printer when required.
*
* Usage: lpon device bits to set
* For example, lpon /dev/cua4 4 to set bit 3 on
* port /dev/cua4.
* 4 = 0100 or bit 3 which is DTR
* 2 = 0010 or bit 2 which is RTS
* 6 = 0110 or both DRT RTS
*/
#include sys/types.h
#include sys/ioctl.h
#include termios.h
#include fcntl.h
#include errno.h
#include stdlib.h
#include unistd.h
#include stdio.h
#include signal.h
#include lpswitch.h
/* Main program. */
int main(int argc, char **argv)
{
struct termios port_config;
int fd;
int set_bits = 6;
/* Open monitor device. */
if ((fd = open(SWDEV, O_RDWR | O_NDELAY)) 0) {
fprintf(stderr, lpswtich: %s: %s\n, SWDEV, sys_errlist[errno]);
exit(1);}
cfmakeraw( port_config );
port_config.c_iflag=port_config.c_iflag|IXON;
port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
tcsetattr( fd, TCSANOW, port_config );
ioctl(fd, TIOCMSET, set_bits );
sleep(5);
close(fd);
}
/* lpswitch.h
* include file for lpswitchd configuration
* (c) 1994, David Cook [EMAIL PROTECTED]
*/
#includetermios.h
#define FILTERDEUG 0 /* filter app debug */
#define DAEMONDEBUG 0 /* daemon app debug */
#define VERSION 0.6 /* appl version number*/
#define LOCKF /var/run/lpswitchd.pid /* lock/PID file */
#define READYFILE /tmp/lpready /* printer ready file */
#define RQSTFILE/tmp/lprequest /* lprequest file */
#define LPDEV /dev/lp0 /* physical lp device */
#define SWDEV /dev/ttyC0/* switch port-HW box */
#define SPEED B9600 /* port baud rate */
#define RESET B0 /* reset by 0 speed */
#define WARMUP 45 /* 45 sec warmup delay*/
#define IDLE1200/* 1200 seconds (20min)
idle delay */
#define XON 17 /* XON character */
#define XOFF19 /* XOFF character */
#define ABORTTIME 90 /* Max before abort */
dbc.
David Cook
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[Asterisk-Users] Re: mysql based adressbook with agi and web interface ?
Quoting [EMAIL PROTECTED]: Subject: [Asterisk-Users] mysql based adressbook with agi and web interface ? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi, I'm looking for adressbook that could easily integrate into Asterisk, so it should: - have agi script to search for caller id name - have fields for notes on previous contacts (for CRM spawning of FOP) - have web interface to edit entries easily ... Any advice, pointers ? What is your favourite address book to use with Asterisk ? I can't speak to the AGI portion - interesting concept - but don't forget interfacing with PIM's and display sets like Cisco 7960. There is an addressbook on the wiki that creates Cisco 7960 address books which I hacked to use the schema of my choice. Then use integrators like Outlook Connect from [EMAIL PROTECTED] to sync office contacts with it. (Sorry gang, have to use Outlook in the corporate world.) Outlook Connect syncs via SSH tunnel to the home server. This gives me consistent context between Palm Treo (Palm syncs to Outlook as part of Hotsync)/Outlook desktop at contract employer/home grown customer management system at home Cisco phones at home. Adding dial-by-name type function would be very cool. If you want the XML hack for the phones you are welcome to it but it's not much other than some modification of the one on the wiki. David Cook ___ 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] Re: Asterisk-Users Digest, Vol 6, Issue 284
1. Create agent phone logins. 2. Create real-time report to monitor agent login/logout activity. Should have the ability to view which agents are currently logged in/out of the system. 3. Create historical report to pull agent activity. Should display login/logout activity. Be able to pull information by rep and timeframe. 4. Create hold calls/bypass statuses for agent login. This status should allow the rep to pause all incoming calls to their login for reasons such as: 1-Break, 2-Lunch, 3-Meeting, 4-Project, 5-Other. This status should not log the agent out of the phone, but only temporarily take them out of the queue to receive the next available call until they end the hold/bypass status and make themselves available for incoming calls. I?m thinking no, but I figured I?d ask anyways before telling my bosses they?re out of their minds. Even if there's an existing interface out there that can provide 1 or 2 of these things, it'd be a nice start. Most of it I'd have to work with a developer to get created, and I'm thinking option 4 is impossible, but 1 2 and 3 is possible with time. Help? Don't tell your bosses they are out of their minds! These features all exist in one form or another on the call centre big iron from companies like Avaya, Nortel, Aspect, etc. If your bosses have experience in larger call centres they will know about these features and it is an extremely mature market. Is Asterisk mature enough to play there ... not yet, but obviously we all hope to get there. Considering the maturity of this market, it would be wise to reuse the nomenclature and process (in principle) that the major players have already done in order to maintain comfort level with our bosses that might let us deploy this thing! #1 is a configuration issue usually done through a control/reporting station like an Avaya CMS. #2 is usually called Real Time Adherance reports. #3 is agent statistics and often is complex enough to be a stand-along package. #4 is called make busy and is one of several states that an agent can be in while present in the queue. Others include things like: - (acd) ready - on acd call - on non acd call - make busy - after call work - on (other media) call/event. etc. FYI The major players have also unbundled portions of their equipment and one of them has even embraced Open Source. Avaya runs their flagship product line on commodity Intel servers on Red Hat linux -- so we are in good company. David Cook ___ 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] Re: Any interest in a Canadian Asterisk
Quoting [EMAIL PROTECTED]: Would it be considered trolling to start a thread on Cleaning Maple Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? Let's not forget the weekly tooques and telephony segment, and a review of the best block heaters for your wi-fi fones. Oh, we're gonna have a good time next Thursday. We need to get Molson Canadian to sponsor us and find Bob Doug for the event? By the way, eh. It's hard to get the moose to cooperate. When you put the parabolic antenna on his antlers you have to ride him backwards when you're leaving your cabin, eh. dbc. -- David Cook ___ 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] Re: Toronto
It looks like this meetup group is becoming the venue for contact as most Toronto respondents have signed up here. Can I request that the remainder who responded to the Toronto call signup? Shidan is the organizer and has proposed a date. I'd hate to confirm it without the remainder having input - especially those with travel times like Andrew K. out in Listowel. Thanks, dbc. -- The system said designed for Windows NT or better. So I installed Linux. Quoting [EMAIL PROTECTED]: Anyone in the Toronto area interested in getting together to share notes and swap war stories? One of the other guys in Toronto interested in * put together a meetup.com group. Please join in and we can see where to go from there. http://opensource.meetup.com/42/ -- David Cook ___ 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] Re: Toronto?
I'm Toronto (well Pickering). I think that could prove helpful. -- David Cook Quoting [EMAIL PROTECTED]: Anyone in the Toronto area interested in getting together to share notes and swap war stories? -- Jim Van Meggelen [EMAIL PROTECTED] ___ 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] Re: FXO to IAX on ethernet. or FXO to SIP on Ethernet
Checkout http://www.mediatrix.com (FXO device 1204) or http://www.multitech.com. I have been looking into this myself. It appears that Nortel has an arrangement with Mediatrix and uses these devices where a remote FXO is needed that would be cost prohibitive to put in a full chassis. Avaya appears to have the same type of arrangement with Avaya where a G700 chassis is overkill. On both fronts I am *assuming* the quality and echo can is excellent if these two players are endorsing this solution. However, they are not in the price range of the products most of us have been using for FXO interfaces on this list. They may not also have the feature versatility we would like in a SOHO environment as their primary market will focus on quality but with dedicated purpose. The Mediatrix is a 4 port FXO only. MultiTech offer more units in different port counts, but each port appears to have flexible config options (FXO/FXS/EM, etc.) which adds significantly to the price. Mediatrix is list price 650.USD and the 2 port MultiTech looks to be 900. USD list. dbc. -- David Cook Quoting [EMAIL PROTECTED]: I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote locations like smaller towns so they would need to be simple, stable and require little to no user intervention after they are installed. Does anyone know of any hardware that will do this or a way that this could be done or ?? Thanks David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there hardware to remote control
From: Ronald Wiplinger [EMAIL PROTECTED] Subject: [Asterisk-Users] Is there hardware to remote control available? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed I am looking for a hardware, which can turn on / off (control) via the dial plan. Is something available? You can run an AGI from within your diaplan which can do anything available to the host machine. As for turning things on/off, you have several options. a) serial port control; b) parallel port control; c) attached microcontroller; d) X-10 signals. Please exuse this for going OT into home automation stuff, but in an effort to answer the original question, here goes ... a) I have often used a little program that flips the DTR RTS signals on a serial port (independently so you can control two things). You need to turn on/off a logic state or an LED that is fine. If you need to switch a larger electical load, put a solid state relay on that pin. I have my laser printer and my pool pump controlled that way. b) Parallel port works basically the same way with the 8 output pins on the connector that can be controlled. Haven't actually done this though. Lastly, connect a microcontroller like a Parallax Basic Stamp to your server where you can write code that runs on the microcontroller and does numerous things pseudo autonomously from c) Microcontroller like the Parallax Basic Stamp series. This allows you to run a program on this little computer device (100.00) that was made for I/O control. It can do all kinds of things pseudo autonomously and feed back the info to the PC. d) X10 have several interfaces for PC's. I like a little one called the Firecracker interface. It uses an RS232C line and can control devices by sending radio signals from it to a reciever module that is plugged into a wall socket. It then embeds the cammands you sent it into the electrical circuits in your home. Another module then plugs into the wall somewhere and you plug devices into it. The little wall modules recieves the signal coming along the electrical lines and turns the device on/off/dim, etc. The reason I like the Firecracker is that it is a dumb device. All program code must exist on the PC therefore I have more control. They have other devices which you download program code to then they are autonomous which I don't think is what you are looking for. I use a) d) extensively here. If anyone wants the code or more info, just ask. David Cook ___ 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] OT: How do I know if I should have IO-APIC?
With regards to the IRQ sharing situation on 400P/X100P cards how would I know if I can use IO-APIC? I am running RHEL 3 on a Dell PowerEdge 1400SC. RHEL installs without IO-APIC support. Is this because RH is overly conservative or because it queried my machine and that is the appropriate option? Does RHEL 3 have a kernel for IO-APIC if appropriate or am I expected to do a custom kernel build to get there from here? dbc. -- David Cook ___ 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] SPA-3000 and distinctive ring
I'm looking to give the SPA-3000 a whirl as I'm having too much difficulty with the irq sharing thing inside the box. I'm reading the book but without having one in-hand to play with it appears a little obtuse at this time. Before I drop down my money can someone with some hands-on with one of these confirm if the SPA-3000 can: a) detect inbound distinctive ring (this looks to me like cfw sel1 caller command from the pdf guide) and if so... b) direct individual distinctive rings to a different Asterisk exten (looks like cfw sel1 dest command to me). ... and if so, does this work reliably enough to be a viable production solution? I presume this means that I can have it ignore other patters I don't want it to pick up at this time (spouse factor) by only specifying certain ring patterns to have a select setting. Thanks! -- David Cook ___ 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: Voicemail shorter then (ex) 2sec - don't accept
On Mon, 2004-11-15 at 11:31 -0500, Seth Remington wrote: On Sun, 2004-11-14 at 13:02, Joseph wrote: In which configuration file I can specify that I don't want to accept messages for example shorter then 2sec. ? I've looked in voicemail.conf but I couldn't find any setting that will support this option. In most cases message shorter then 2 or 3sec will not contain any message and I don't want system to record them and sending an email to me. You were looking in the right config file. The parameter is called maxmessage. -Seth I just checked and I think this is not the one. maxmessage is to limit the message to the amount of time you specify in seconds. What I was looking for was to discard all the messages that are 3sec. or shorter. -- #Joseph Also a while back I noticed it did not understand that a message length equal to that of maxsilence was a null message and to discard it. David Cook ___ 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: Geotel integration with Asterisk
Geotel is a company that Cisco bought which provides call control across geographically dispersed locations. The simplest application is being able to query call queue status at another location. For example, a call comes in and can be sent to one of three call center locations. Geotel can query each location to see who is the least busy for this type of call. Traditionally it has been VERY expensive. We provide some primitive Geotel functions in-the-cloud right now. For example, we can know how many live calls are going to a location before we send the call. We can set thresholds (e.g. if a location A has over 100 concurrent calls send them to location B). Geotel can theoretically provide this and carry it further. I think there is some nice enterprise reporting that can come from the Geotel as well. G. Their greatest claim to fame is that their peripheral monitor PC sits on your premise, and connects to your brand x pbx to report upstream to the telco router (actually a redundant pair PC) as to the ingoings of your call centre. The decision to terminate the call on a particular call centre is done in the telco cloud at the SS7 layer. Each call centre has 250ms to respond to the correct status or the telco default-routes the call based on the tables in the NAM. This feature is self-healing dynamic routing. Proactive rather than reactive when your call volumes change or a failure takes a centre offline/snow storm means only half of your agents show up today in one area of the country, etc. It allows a translation between disparate PBX's to participate in this scheme so it is a huge boon in mergers/acquisitions. Just drop this Peripheral Monitor (pair) in your CC and you are intergrated into our enterprise. Actually reporting is one of the weakest links in the Geotel (now Cisco ICM (Intelligent Call Manager)) platform. Countless clients complain about this and at their user conference they even came out and admitted it. The data elements are there, but they don't have a good handle on how to rationalize them. Bell Canada, Allstream, MCI and ATT offer this now that I am aware of. Yes it is very expensive, but for multi-site high-availability services like banks, airlines and insurance companies it pays off in spades. dbc. -- David Cook ___ 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: Multi-office topology suggestions
We have five offices in Canada. Our main office is in Edmonton , with branch offices all over the nation. I would like to place the Asterisk server in the Edmonton office and have it route calls to the branch offices. I would also like to have each of the branch offices have a local phone number. That local phone # would actually dial into the Asterisk box , and then routed appropriately via VPN to the correct location. This gives us a method of controlling and tracking all calls made to all offices. Several ways to skin that cat. If you prefer/need one switch, then you will probably want remote FXO in the smaller offices. Something like a Sipura SPA-3000 at the low end or moving up to some of the larger multi-fxo devices that are out there. This will put a huge importance on the quality/availability of your data circuits and provider. In Canada you have Telus in the west, Bell in central and Alliant on the coast. Allstream is smaller but nation-wide and concentrates on business. Depending on where you are (on-net/off-net) you will likely be in a wholesale market at one location or the other and this will by definition hurt your Mean Time To Repair (MTTR). Or you can have an asterisk server at each location and combine the dial plan. Depending on your calling patterns, you probably want to write some custome Least Cost Routing code (agi can work for this) to determine if it's better to trunk the call from A to B and pickup PSTN from there, or just call PSTN from your local site. You might want to have the same voice menu at each location for availability rather than have everything trunk back to one. That way, you wouldn't lose your menus if you had a network outage. You will also want to ensure the LCR code accomodates dead IP paths and automatically routes over PSTN to get there seamlessly to employees customers. (Inbound call from site A to site B is recognized by ANI and behaves like it was an internal call trunked over IP, etc.) This way your employee sat and customer experience don't change and you just modify your phone bills dependant on how good your data providers are. dbc. ___ 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: CTI development
Or what is it that you meant in particular? I'l bet he means 3rd party call control like in a traditional CTI deployment ala Cisco ICM, Genesys or an oldie-but-goodie, IBM CallPath DirectTalk. (Net-net version) Basically, a scratch-pad type area of ~2K that gets created/destroyed with every call and _follows_ the call for its life in the system. Olus the ability of a 3rd party computer application aka softphone to control the telephone appliation - this part we've got but still needs some modification for true CTI. (Example) So the caller gets to the IVR. The IVR pushes data relevant to the current call onto the scratch pad using a unique call event ID then xfers the call to the call centre Q. The call gets allocated to an agent in the Q. Their desktop application gets an alerting message which is basically a ring event alerting them that they are about to get the next event including the internal ID of the event. (In traditional environments this happens _slightly_ before the phone rings. The application then reads the scratch pad data associated with the call event ID so the desktop can have full context of what has gone before in the call. The desktop application then does whatever it needs to do in the customer environment - this is custom development - the CTI vendor offers an SDK for interface to their softphone product. The desktop application needs the ability to also write/update to the scratch pad as there may be a need to xfer the call to another agent or back to the IVR which should be able to read the updated data. I may not have the skill to code all of the application, but I'm a call centre solution architect. If anyone would like to bring this functionality to Asterisk I would be excited to offer industry advice. There are lots of gotchas in the CTI world that are completely _not_ related to programming skill. The wrong implementation simply won't have a market. dbc. -- David Cook ___ 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 strategy for echo cancellation.
I'd like a good plan for this too, however this problem seems to exist only with analog FXO interfaces. If you have 12 lines, would it not have been cost effective to go fractional T1 then the box would be cleaner and the problem be averted? Quoting [EMAIL PROTECTED]: I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office, using three TDM400's with 4 FXO's each for incoming calls. Outgoing calls are (for the moment) routed via VoicePulse. ___ 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: Distinctive Ring
Quoting [EMAIL PROTECTED]:
If I have a Wildcard X100P and Asterisk, it is possible to make it
answer only the distinctive ring call of two short rings and ignore
the regular incoming ring?
Bill Lohr
Absolutely.
[zapata.conf]
dring1=0,0,0
dring1context=distring1
dring2=326,0,0
dring2context=distring2
dring3=93,0,0
dring3context=distring3
dring4=94,0,0
dring4context=distring4
channel = 1
[extensions.conf]
[distring1]
exten = s,1,NoOp(${CALLERID})
[distring2]
exten = s,1,NoOp(${CALLERID})
[distring3]
exten = s,1,NoOp(${CALLERID})
[distring4]
exten = s,1,NoOp(${CALLERID})
Each context distring1-4 will be answered by the appropriate distinctive
ring cadence. The NoOp command will log the CALLERID to the CDR
database so you still have a record of it.
Simply put your real extensions in the context that you want answered
in place of the NoOp command.
In zapata.conf you will need to change the dringx line to be the three
digit code that shows in your console when that ring cadence arrives.
So you need to call the system with that number and record the values
you get on the console, then put them in zapata.conf as appropriate.
Enjoy.
dbc.
--
David Cook
This message was sent using IMP, the Internet Messaging Program.
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[Asterisk-Users] Re: Distinctive ring detection problem
Sure, this one works. You need a dringX definitions of the distinctive rings. Put in each one the output you get in the log for the call pattern when the phone gets answered. [channels] switchtype=national signalling=fxs_ks usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=256 echocancelwhenbridged=yes echotraining=yes rxgain=-5.0 txgain=-5.5 group=1 callgroup=1 pickupgroup=1 immediate=no dring1=0,0,0 ---from the log output when phone answered. dring1context=advan-mainline dring2=326,0,0 ---from the log output dring2context=advan-fax dring3=93,0,0 ---from the log output dring3context=distring3 dring4=94,0,0 ---from the log output dring4context=distring4 Quoting Paul Budden [EMAIL PROTECTED] I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default usedistinctiveringdetection=yes dring1=134,0,0 dring2=137,0,0 dring1context=internal2 dring2context=default channel = 1 -- David Cook ___ 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] Still unacceptable echo on X101P
I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not having the system pickup/handle all the calls in the house until I'm ready to launch it. 2. What about the effects of it being downstream from a DSL line filter? 3. If impendance mismatch is the (or a major contributing) factor, can we not devise some interface circuit which will allow a variable rate on the impedance so we can dial out the echo based on individual line conditions? dbc. -- David Cook ___ 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: Overhead Paging
Quoting Brian Pavane: My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an -Brian Not sure what a Paging Unit is. Some kind of auto-answer phone with audio outputs?? I just used the sound card in the PC plugged into an amplifier. Haven't seen any detrimental effects using the local processing power for this. [paging] ; Overhead paging through the sound card exten = 2900,1,Ringing exten = 2900,2,Dial,console/dsp exten = 2900,3,Hangup -- David Cook ___ 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: Re: 2 servers
Quoting [EMAIL PROTECTED]:
From: Kanuri, Seshu [EMAIL PROTECTED]
Dave,
I am implementing this solution and would appreciate if you can send
me the doc at this email address - [EMAIL PROTECTED]
Thanks
Seshu Kanuri
Enough people have asked me for this that I will try and condense it for
the list. I admit I wanted to put it on the wiki and couldn't figure
out how to start a new page!!! (Maybe I'm just thick ;-(
There is also another document on the wiki about the subject at
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
Anyhow, here is mine:
Method 1
Rec'g Svr
iax.conf
[REC_SERVER]
type=user
host=my.calling.server.ca
secret=mysecret
context=local
trunk=yes
Send'g Svr
extensions.conf
[mycontext]
exten =
_5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _5XXX,2,Hangup
exten = _5XXX,102,Hangup
Any call in the mycontext context on Calling Server to extensions
5000-5999 (mapped by extension _5XXX) will get sent to receiving server
(my.receiving.server.ca) into the local context on the receiving
server.
Performing the same configuration in the opposite direction will allow
cross-calls between Asterisk systems.
Pros:
Simple, all references in one file per server.
Cons:
Information in dialing string will appear in logs inclusive of
user:password.
Dial string becomes very long.
Method 2
Rev'g Svr
iax.conf[REC_SERVER]
type=user
host=my.calling.server.ca
secret=mysecret
context=local
trunk=yes
Send'g Svr
iax.conf
[REMOTE_SERVER]
type=peer
host=my.receiving.server.ca
secret=mysecret
context=local
extensions.conf
[mycontext]
exten = 5XXX,1,Dial(IAX2/REMOTE_SERVER/${EXTEN})
exten = _5XXX,2,Hangup
exten = _5XXX,102,Hangup
Pros:
User:Password are stored in the calling server?s iax.conf file and not
part of the Dial string. This is more secure in that they are not
recorded in log in files.Dial strings much shorter and concise.
Cons:
Calling server now must have iax.conf and extensions.conf coordinated
making setup a little more complicated.Must user ?type=? definition
correctly:Caller = ?peer?; Receiver = ?user?Type=friend is a
bi-directional relationship meaning both ?peer? and ?user? at the same
time.
Unknown IP (Dynamic IP on one server)
Register Command
If the calling server does not have a fixed IP address or DNS namespace
then the iax.conf file description of the calling server located on the
receiving server should specify host=dynamic.
If the calling server host is specified as dynamic, the calling server
must register with the receiving server with the register command.
Rec'g Svr
iax.conf
[REC_SERVER]
type=user
host=dynamic
secret=mysecret
context=local
trunk=yes
Send'g Svr
iax.conf
register = REC_SERVER:[EMAIL PROTECTED]
[REMOTE_SERVER]
type=peer
host=dynamic
context=local
extensions.conf
[mycontext]
exten = 5XXX,1,Dial(IAX2/REMOTE_SERVER/${EXTEN})
exten = _5XXX,2,Hangup
exten = _5XXX,102,Hangup
I hope this is both accurate and helpful!
dbc.
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[Asterisk-Users] Re: 2 servers
Quoting [EMAIL PROTECTED]: How do I get town A people to dial 201 and it will go to sown B's server's 201 SIP users Please not that I'm only a newbie and my terms may be wrong but I'm really having a bod time with this Please help Thanks ALtus I have a doc on it. (Sorry was going to copy/paste but my mail reader didn't like the columns from the doc.) If you want it drop me a line and I'll send you the file. (Should also probably put it in the wiki :-) dbc. -- David Cook ___ 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] (no subject)
The wiki page on Asterisk + Nat (http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions) lists the possible types of server/client relationships with one most probably interesting to us being #3. snip 3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk snip Then it goes on to say: * #3 Works with port forwarding and some header mangling magic Can somebody explain a little more about the header mangling magic as it is not discussed anywhere else in the document. Currently I have my firewall port forwarding 5060 to my asterisk server and the UDP port range forwarded as well. Registration works, but no audio. Obviously the RTP stuff is not happy with the forwarding. dbc. -- David Cook ___ 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] Lost 7960 time display on upgrade
I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 172.16.10.24 ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: EST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: April ; Month in which DST starts dst_start_day:; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: ; Day of month in which DST stops dst_stop_day_of_week: Sunday; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) Thanks, dbc. -- David Cook ___ 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: Lost 7960 time display on upgrade
Quoting Rich Adamson On Fri, 13 Aug 2004 06:34:26 -0400, David Cook [EMAIL PROTECTED] wrote: I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Funny enough my phone has done the same thing. I figured it was just a configuration error on my part and haven't had the time to really do anything about it. Hopefully someone will chime in with a solution (assuming its not just a configuration issue). There was an open cisco item on DND and NTP interaction, and have been several issues relating to DNS. If your ntp definition uses DNS, I'd suggest changing to an IP address. I've running v7.1 and have not ever had an ntp display issue. Rich Nope, IP addr used in the NTP config. Does anyone else use SIP 6.3 on their 7960 and the time display works for them? Can you supply your SIPDefault.cnf and SIPxx.cnf's? -- David Cook ___ 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] X100P outbound only (Don't answer)
I tried implementing my * and it didn't pass the spouse factor at this
time. I wanted to hook it up for outbound only at this point to get a
better handle on the dial plans and the echo problem.
I thought this might have been done before as a natural part of testing
- but maybe not.
In wcfxo.c I found this:
if (!wc-offhook !wc-ringdebounce) {
if (!wc-ring (wc-pegcount PEGCOUNT)) {
/* It's ringing */
if (debug)
printk(RING!\n);
zt_hooksig(wc-chan, ZT_RXSIG_RING);
wc-ring = 1;
}
if (wc-ring !wc-pegcount) {
/* No more ring */
if (debug)
printk(NO RING!\n);
zt_hooksig(wc-chan, ZT_RXSIG_OFFHOOK);
wc-ring = 0;
}
}
Is changing the wc-ring = 1 to 0 an appropriate place to fix this for
outbound-only operation?
dbc.
--
David Cook
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[Asterisk-Users] Re: X100P outbound only (Don't answer)
Quoting From: Soren Rathje [EMAIL PROTECTED] No Wait() or Answer() so the line will never be answered but incoming = callerid will be in the log/cdr... :-) /Soren I think I just missed something very fundamental. You are saying that the switch doesn't pickup the PSTN line until one of the choosen destinations performs an action like answer/dial, etc? I thought the switch picked up first, then routed the call based on the dial plan. So I can set usedistinctivering=yes with only an answer disposition/context on dring2 causing * to only pickup if you call that number!!! Very cool. -- David Cook ___ 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: Newbie Questions
I'm going to be helping to set * up for the company I work for, and in doing all my research about it, have found it to be a very viable solution for my SOHO side business at home. I do however have a few questions, forgive me if they're stupid but I'm new to all of this. OK. I'll bite. 1. I want to be able to handle 3 analogue phone lines, with a regular bell telephone line coming into the house. In phone lingo the phone sets are the station side of the PBX. The lines coming into the house are the trunk side. Sometimes, but not often, the station side is called line-side but usually to referrence interconnecting tie lines. three FXS ports and one FXO port? Correct. can I 'chain' my phones together from the one FXS port ... Yes, but they will all be the same extension just like all the phones in your house and not be individually addressable (dialable). upgrade to VOIP capabilities for my SOHO Long Distance, is this as simple as getting another card with a T1 interface ... No. A T1 interface that goes directly into the PBX (Asterisk) is usually for voice (23B+D (23 bearer channels for voice + one data channel for signalling)). You will most likely already have a 100BT connection on your server and that is where you will get the most cost effective in/out IP connectivity to your box. You will then connect your network - or that segment of your network - to the outside IP world either directly or via firewalls/routers. Does * support 'ring tone identification' ? Yes. Relating back to the splitting of the phone lines,... See above. Matt, you really need to spend _a_lot_ of time reading the documentation and playing with the system. There is no substitute for hands-on experience. I have had a long history in data and telephony and I still played with the product for 4 months before I asked a question. Until you spend that amount of time learning you will not have the background to understand the answers that people give you. Most people on the list won't answer a question like this one because it has been well shown that they are wasting their time teaching someone who is not ready for it yet. The other side of the coin is that the people on this list that have spend copious months of their time gaining expertise are perfectly willing to support peers who have the invested in the same manner. They are not however, willing to spoon feed people who have not yet, or appear unwilling to make that investment themselves. Those people need to hire consultants. If you do want to hire a consultant - which there is nothing wrong with do so - just ask for such on the list and there will be manny people willing to provide rates for their services. I know that email is a cold medium and this may come across badly at first, but that does in fact represent the culture of a user community. You need to read up first to gain a minimum level of expertise _as_a_user_ in order to productively take part in the user community. Hopefully I'm clear on my questions, Thanks a lot in advance. Matt Gibson Unix Administrator Experthost / NJ Tech Solutions -- David Cook ___ 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: sendmail.cf and relaymail to a smtp server
I am presuming your Asterisk box is behind your ISP. You don't actually need user/pw to send somebody email in the outside world, but your ISP has prevented you from _directly_ sending email to anybody and make you go through their SMTP server which forces you to authenticate with it like a Mail User Agent rather than a Mail Transport Agent. You don't need relaying, you need to define your ISP as a Smart Host for SMTP and then you need to invoke authentication. Try this link for the auth stuff: http://www.sendmail.org/~ca/email/auth.html Then in your sendmail.cf file you need to specify your ISP as a smart host by looking for the DS line and adding your isp with no spaces like this: DSmyisp.com It is probably too much to delve into at this point, but after you get this working, go check out managing sendmail with the m4 preprocessor. It will allow you to automatically generate sendmail.cf files from a (more) comprehensible file than the .cf. It will make a world of difference once you start chaning more than just one value like we are doing here. dbc. Hello, Who can help me I am trying to setup the sendmail so that I can mail the voicemail's to an internet SMTP mail server. I know that I have to setup the sendmail.cf and configured a relay to my normal SMTP server. I am running RedHat 9 and my internet provider has a SMTP mail server with user and password authentication. Regards, Han ___ 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] X100P bad sound after period of time
Hi folks. I am using a X100P card and after some random amount of time of correct operation, say 8-20 hours, the card starts acting up and producng horrid sound quality which is all broken up. All other channels appear to work fine. One thing I noticed, is that zap show channel 1 always shows the Actual Hookstate: Offhook as soon as the telco line is plugged in. Is this normal? Maybe a bug in the status program or might this be indicative of my problem somewhere? The card claims to be sharing an interrupt with the SCSI controller and I don't see any way to change that. I put a second card in a different machine and it too, shared the interrupt but with the usb-uhci instead. It too shows zap show channel 1 as offhook as soon as the line is plugged in. Is there something real basic I am missing here? I'm on CVS-HEAD-06/27/04-23:21:33 /proc/interrupts CPU0 0: 541808 XT-PIC timer 1: 1203 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-ohci 8: 1 XT-PIC rtc 10:5281916 XT-PIC aic7xxx, wcfxo 11: 22266 XT-PIC aic7xxx, eth0, Cyclom-Y 12: 32 XT-PIC PS/2 Mouse 14: 0 XT-PIC ide0 NMI: 0 ERR: 0 My modules.conf looks like: alias eth0 e100 alias scsi_hostadapter aic7xxx alias usb-controller usb-ohci options torisa base=0xd alias char-major-196 torisa options wcfxo debug=1 options torisa debug=1 options wcfxs debug=1 options zaptel debug=1 zaptel.conf fxsks=1 loadzone = us defaultzone=us zapata.conf [trunkgroups] [channels] context=demo signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context = PSTN-in channel = 1 Thanks, dbc. -- David Cook ___ 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: tdm400p static - out of ideas (Jorge Mendoza)
Ryan, from the console what does zap show channel 1 or 2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: On July 8, 2004 03:22 am, Nicholas Bachmann wrote: Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. Let's look at some possibilities of line problems: What time does it stop answering? Is it ever during ALIT times (usually very early morning)? It's totally random - morning/evening/afternoon. Once it stops answering, that's it, a reboot or module-reload is needed. If ALIT for some reason prevents the card from answering, it should be able to recover and begin answering after the ALIT is complete. Have you tried calling the telco to see if it could be their problem? When the card goes into the non-functional state, I can plug a regular phone into any of the lines and make calls just fine. After verifying working lines and plugging them back into the tdm400p card, I still can't dial out (the Zap channel will answer, but I will hear only static, and the call to the pstn is never placed). As well, incoming calls will not be answered (* console will not even show the 'started simple switch on zap/x' message). How far away from the CO/mux are you? Not too sure - it's in downtown Calgary - so probably not far. There is the possibility that _something_ with the phone line is triggering the problem. Maybe it's some noise, an unexpected signal, some crosstalk ... things that will cause unexpected behavior ... but also things that shouldn't put the entire card into a non-functioning state. Have you tried a new/different card? If you didn't want to fork out the cash for a new one, you could try a X100P/knockoff* on one of the lines to see if that fixes the problem... if so you can deduce a bad card. I may have to push for a replacement tdm400p card from Digium. Nick *I usually don't recommend the knockoffs, but for a day of testing $10 sure beats $100... everybody else should support Digium! :-) An acquaintance who is having the same problem has reluctantly replaced his card with an openline4. I would like nothing more than to stick with Digium hardware - this thread and obtaining a replacement card is my last kick at the cat. -- David Cook ___ 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] SPA-2000 and time of day
Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour slow. I have ntp running on my network and it has been told to respect daylight savings time. Is the SPA omitting this feature? -- David Cook ___ 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] IAX2 Trunking help!
I was just trying to solve this one myself. I found this method worked for me. I'm still calling this Method 1 in my document because I don't fully understand the switch and the register versions and pros/cons to implementation of each. But this one does work. Method 1 Receiving Server Iax.conf [REC_SERVER] type=user host=my.calling.server.ca secret=mysecret context=local trunk=yes Calling Server Extensions.conf [mycontext] exten = _5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _5XXX,2,Hangup exten = _5XXX,102,Hangup Any call in the mycontext context on Calling Server to extensions 5000-5999 (mapped by extension _5XXX) will get sent to receiving server (my.receiving.server.ca) into the local context on the receiving server. Performing the same configuration in the opposite direction will allow cross-calls between Asterisk systems. -- David Cook ___ 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] IAX2 Trunking help!
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten = _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the type=peer/user/friend thing? I did read the docs
but maybe I'm thick. Maybe the visual person in me needs to see a matrix.
Further, If I can get two boxes to talk together like this, what exactly
is the register for ... what does it actually do?
dbc.
Quoting Kevin Walsh [EMAIL PROTECTED]:
David Cook [EMAIL PROTECTED] wrote:
[mycontext]
exten =
_5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _5XXX,2,Hangup exten = _5XXX,102,Hangup
You really don't want your username and password to appear (in
plain
text) in your logs.
Put the sensitive details in iax.conf instead of extensions.conf.
As well as being more secure, it'll make your Dial() string
shorter,
and will mean that you only have to change the connection details
in
one place, should the need arise in the future.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s
h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
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RE: [Asterisk-Users] IAX2 Trunking help!
Perfect! Thanks for the clarification. That's what my brain needed - on both points. dbc. Quoting Kevin Walsh [EMAIL PROTECTED]: If that's on your outgoing side then you'll also need type = peer in there. The incoming side would have type = user. Outgoing = peer, incoming = user. Friend is both incoming and outgoing, but you probably don't want to use that. The register is so that you can use host = dynamic on the incoming side. In that case, you will have to register your location with your peer before you can receive incoming calls. ___ 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
