[asterisk-users] Planning 48 Station Install, Need advice on several topics
I'm planning a new * system which will utilize 48 stations (Polycom Soundpoint 501s mostly) and a dual span PRI card and I have some questions. The system will host MeetMe conferences of 10-15 users on a regular basis and see fairly high usage as it is going into a medical setting. 1. I haven't built a system this big before, will a processor such as the Intel Pentium D 830 3.0GHz / 2MB Cache / 800 FSB / Dual-Core be sufficient for the task? If not, what should I be considering? Also, the system is to have a dual span T1 card such as the Digium T205P or the T207P. One spam will connect to the PSTN while the other span will connect to a MultiTech RAS server. The idea is to look at an inbound call's extension and if it is a data call for the MultiTech, then dial the MultiTech's trunk and pass the data call through. 2. Is there anything inherently wrong with my line of thinking here? 3. Is the Digium dual span card the one to go with here and is the onboard echo cancellation better or worth the money? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Panasonic Hybrid Integration Advice Needed
I have a client who has a Panasonic Hybrid system. They are taking in another company as a building tenant and the tenant will be on a new 12 station Asterisk system. This new asterisk system will have 4 FXO ports plus ITSP. The two systems will be separate except that they should tie together for the purposes of dialing extensions directly on the opposite phone system and for transferring calls. I'm looking for advice on how best to accomodate this. Is it possible to do this via the Panasonic's IP interface or will I need to cross connect them via T1 cards? This is my first integration as you can probably surmise. Thanks in advance. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 and CDR
I have the same problem. Please reply to the list if you figure it out. I'll do the same. _ From: Pablo Almido [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 17, 2007 9:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4 and CDR Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file cdr_addon_mysql.so Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --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] TDM400P, fxotune and ADSL filters - Just a FYI, FWIW
This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c and fxotune.h from cvs and recompiled zaptel. fxotune then ran but I got the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! After two days I installed a splitter to listen in and found out that fxotune wanted 18 seconds of silence on the line but Bellsouth only gives 15 seconds. The -m switch in ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 allowed the test to complete successfully. Before tuning the TDM400P with ./fxotune -s, I observed the echo percentage on the line with ./fxotune -d -b 4 to be .32, a far cry from the .05 I wanted. After ./fxotune -s, ./fxotune -d -b 4 revealed an echo percentage of .075, still not good enough. I remembered that there is a DSL filter between this FXO module and the PSTN to break out signal for my DSL modem. I removed it and plugged the FXO straight in to PSTN. After a rerun of ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 and a ./fxotune -s, ./fxotune -d -b 4 now reveals .026 percent echo! It appears that the DSL filter circuitry affects the .fxotune impedance test to the point that it becomes ineffective (~.05 delta in my case) FWIW, I replaced the filter and reran ./fxotune -d -b 4 and observed a report of .11 percent echo, which I do not trust due to the filter's effect on the circuit. I eagerly removed the aggressive suppression and restored the original echo canceller to be disappointed that the echo still exists. So it is back to Mark2 with Aggressive. If you hang a FXO module behind a DSL filter and have high echo percentages or echo, this is a gotcha. I'm now experimenting with zaptel 1.4 with similar results, despite a new default echo algorithm. Also, any tips on echo reduction from here would be greatly appreciated, I'm out of ideas. My biggest fear is installing a hybrid system in a client's office and to come across a situation where I can't suppress echo.. ___ --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] TDM400P, fxotune and ADSL filters - Just a FYI, FWIW
Eureka, echo free at last! ahh I set the rxgain by running my CO's milliwatt test to 14844 from the original 6688. I just looped from FXO 3 to the asterisk milliwatt() test on FXO 4, I just found the txgain was 6686, instead of 14844. (http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.h tml) After bumping the txgain to 6 (!), I got it to 13500 and that was all I could get. However, The echo has disappeared. Sorry to answer a question that hasn't been asked, but maybe this will save someone some serious frustration! _ From: John French Sent: Monday, January 15, 2007 11:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM400P, fxotune and ADSL filters - Just a FYI, FWIW This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c and fxotune.h from cvs and recompiled zaptel. fxotune then ran but I got the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! After two days I installed a splitter to listen in and found out that fxotune wanted 18 seconds of silence on the line but Bellsouth only gives 15 seconds. The -m switch in ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 allowed the test to complete successfully. Before tuning the TDM400P with ./fxotune -s, I observed the echo percentage on the line with ./fxotune -d -b 4 to be .32, a far cry from the .05 I wanted. After ./fxotune -s, ./fxotune -d -b 4 revealed an echo percentage of .075, still not good enough. I remembered that there is a DSL filter between this FXO module and the PSTN to break out signal for my DSL modem. I removed it and plugged the FXO straight in to PSTN. After a rerun of ./fxotune -i -m 15 -vv -b 3 -e 4 -t 2 and a ./fxotune -s, ./fxotune -d -b 4 now reveals .026 percent echo! It appears that the DSL filter circuitry affects the .fxotune impedance test to the point that it becomes ineffective (~.05 delta in my case) FWIW, I replaced the filter and reran ./fxotune -d -b 4 and observed a report of .11 percent echo, which I do not trust due to the filter's effect on the circuit. I eagerly removed the aggressive suppression and restored the original echo canceller to be disappointed that the echo still exists. So it is back to Mark2 with Aggressive. If you hang a FXO module behind a DSL filter and have high echo percentages or echo, this is a gotcha. I'm now experimenting with zaptel 1.4 with similar results, despite a new default echo algorithm. Also, any tips on echo reduction from here would be greatly appreciated, I'm out of ideas. My biggest fear is installing a hybrid system in a client's office and to come across a situation where I can't suppress echo.. ___ --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] fxotune Error - Found a solution in my case
I finally isolated the problem, hope it works for you too. The default -m (silencegoodfor) default of 18 seconds is too long for my telco and the test is getting interrupted. I listened in on the line with a splitter and realized what was happening. I had to set -m down to 15 seconds in my case and it works. Also, I'm running zaptel-1.2.12 so I had to svn checkout the current zaptel trunk to get a better version on fxotune as the one in 1.2.12 has very few options. I still have some echo but at least I'm on the right path now. Also anyone have any tips on plotting the -d values? - Original Message - From: Gordon Henderson [EMAIL PROTECTED] Sent: Sun, 1/14/2007 3:28am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] fxotune Error On Sat, 13 Jan 2007, John French wrote: I'm trying to learn to use fxotune and am getting the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! when I run it. This is a new install of Asterisk 1.4 and zaptel 1.4. Also I can't really find an up to date how to on it. Am I supposed to place fxotune on my startup path and run fxotune -s once it is working correctly. Any nuggets of info are most appreciated. This isn't any help, but I've just seen this myself on one of my units. Hardware wise it's indentical to a dozen other units I've built, and it does work OK, I just get the same error as you when trying to fxotune it. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possibleto use zaptel 1.4 with asterisk 1.2?
Is it possible to use zaptel 1.4 with asterisk 1.2? I'd like to try to resolve some echo issues but have some applications which aren't yet compatible with asterisk 1.4 ___ --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] fxotune Error
I'm trying to learn to use fxotune and am getting the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! when I run it. This is a new install of Asterisk 1.4 and zaptel 1.4. Also I can't really find an up to date how to on it. Am I supposed to place fxotune on my startup path and run fxotune -s once it is working correctly. Any nuggets of info are most appreciated. [EMAIL PROTECTED] zaptel-1.4.0]# ./fxotune -vv -i -b 3 -e 4 Tuning module /dev/zap/3 Resetting line 0,0,0,0,0,0,0,0,0,-1,1998.667009,0.422893 10,0,6,1,254,2,255,0,0,-1,1472.309836,0.311523 3,255,255,0,1,0,0,0,0,-1,930.662538,0.196917 3,1,253,253,2,255,0,0,0,-1,753.304640,0.159390 9,254,251,255,2,0,1,0,0,-1,463.526511,0.098076 5,3,251,250,2,254,0,0,255,-1,262.730838,0.055591 8,253,2,244,255,10,244,3,253,-1,385.563899,0.081581 10,249,244,8,12,245,252,0,1,-1,968.228623,0.204865 1,0,0,0,0,0,0,0,0,-1,1809.818403,0.382935 10,252,255,1,255,0,0,0,0,-1,1050.062881,0.222180 7,255,251,251,2,255,255,1,255,-1,694.337920,0.146913 3,1,251,250,1,254,255,0,255,-1,390.528753,0.082631 5,252,250,0,0,255,1,0,0,-1,229.706450,0.048603 5,3,251,250,1,253,0,0,255,-1,276.796628,0.058567 8,253,2,244,255,10,244,3,253,-1,385.488193,0.081565 10,249,244,8,12,245,252,0,1,-1,968.380696,0.204897 2,0,0,0,0,0,0,0,0,-1,927.067067,0.196156 7,0,0,255,254,0,0,0,0,-1,865.567321,0.183143 9,0,253,254,2,255,0,0,0,-1,417.253736,0.088286 5,1,249,254,4,253,1,0,0,-1,288.752351,0.061096 5,252,250,1,1,254,0,255,0,-1,223.330159,0.047254 5,3,251,250,2,253,255,255,255,-1,355.908340,0.075306 8,253,2,244,255,10,244,3,253,-1,385.646453,0.081598 10,249,244,8,12,245,252,0,1,-1,967.460662,0.204703 3,0,0,0,0,0,0,0,0,-1,937.031070,0.198264 7,0,255,254,255,0,255,0,0,-1,797.578151,0.168758 9,0,253,253,1,255,0,0,0,-1,314.018726,0.066442 5,1,249,254,3,253,1,0,0,-1,241.203980,0.051036 5,252,250,1,1,254,0,255,0,-1,223.249069,0.047237 5,3,251,251,2,253,255,255,255,-1,375.524487,0.079456 Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! Tuning module /dev/zap/4 Resetting line 0,0,0,0,0,0,0,0,0,-1,1570.495018,0.332297 10,0,6,1,254,2,255,0,0,-1,1357.698346,0.287272 3,255,255,0,1,0,0,0,0,-1,897.370834,0.189873 3,1,253,253,2,255,0,0,0,-1,826.115747,0.174796 9,254,251,255,2,0,1,0,0,-1,801.017936,0.169485 5,3,251,250,2,254,0,0,255,-1,517.786442,0.109557 8,253,2,244,255,10,244,3,253,-1,596.448176,0.126201 10,249,244,8,12,245,252,0,1,-1,1334.651950,0.282396 1,0,0,0,0,0,0,0,0,-1,1584.490976,0.335259 10,252,255,1,255,0,0,0,0,-1,1313.891325,0.278003 7,255,251,251,2,255,255,1,255,-1,1027.128645,0.217328 3,1,251,250,1,254,255,0,255,-1,782.937418,0.165660 5,252,250,0,0,255,1,0,0,-1,785.872695,0.166281 5,3,251,250,1,253,0,0,255,-1,570.158387,0.120638 8,253,2,244,255,10,244,3,253,-1,597.086203,0.126336 10,249,244,8,12,245,252,0,1,-1,1334.956109,0.282460 2,0,0,0,0,0,0,0,0,-1,760.888726,0.160995 7,0,0,255,254,0,0,0,0,-1,870.134357,0.184110 9,0,253,254,2,255,0,0,0,-1,591.725939,0.125202 5,1,249,254,4,253,1,0,0,-1,573.417242,0.121328 5,252,250,1,1,254,0,255,0,-1,676.772025,0.143197 5,3,251,250,2,253,255,255,255,-1,522.890543,0.110637 8,253,2,244,255,10,244,3,253,-1,596.466272,0.126205 10,249,244,8,12,245,252,0,1,-1,1334.947255,0.282458 3,0,0,0,0,0,0,0,0,-1,836.837196,0.177064 7,0,255,254,255,0,255,0,0,-1,854.068171,0.180710 9,0,253,253,1,255,0,0,0,-1,616.414741,0.130426 5,1,249,254,3,253,1,0,0,-1,587.765384,0.124364 5,252,250,1,1,254,0,255,0,-1,676.604040,0.143161 5,3,251,251,2,253,255,255,255,-1,471.160185,0.099692 Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! Unable to tune 2 devices, even though those devices are present [EMAIL PROTECTED] zaptel-1.4.0]# ___ --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] Need help getting anterisk application to load
This is probably a newbie question: I'm trying to get the asterisk weather app at http://nerdvittles.com/index.php?p=136 installed on Centos 4.2 x86_64 SMP ( Linux localhost.localdomain 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:28:26 CDT 2006 x86_64 x86_64 x86_64 GNU/Linux ) rpm -ihv flite-1.3-1.aah.i386.rpm seems to complete successfully rpm -ihv app_flite-0.3-1.aah.i386.rpm seems to complete successfully Restarting asterisk with asterisk sv reveals: [app_flite.so]Jan 11 08:40:17 WARNING[10227]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_flite.so: cannot open shared object file: No such file or directory Jan 11 08:40:17 WARNING[10227]: loader.c:554 load_modules: Loading module app_flite.so failed! But it is there: [EMAIL PROTECTED] ~]# ls /usr/lib/asterisk/modules/|grep flite app_flite.so I've never used x86 before is this a compatiblity issue with the 64 bit OS? How can I get this running? Any help is appreciated! I do have this app running just fine on FC4 i386 on another computer. ___ --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] MeetMe() not recording calls
When I try to record a call the console shows: www*CLI Starting recording of MeetMe Conference 1 into file meetme-conf-rec-1-1167836078.0.wav. www*CLI The code being executed in extensions.conf is: exten = s,n(record),MeetMe(,rDMpc) ;Make new Room and record call. exten = s,n(bye),Playback(vm-goodbye) exten = s,n,Hangup The file never appears in /var/spool/asterisk/meetme Installed sw is: asterisk-1.2.14 kernel-2.6.18-1.2200.fc5.src.rpm asterisk-1.2.14.tar.gz lame-3.96.1 asterisk-addons-1.2.5lame-3.96.1.tar.gz asterisk-addons-1.2-current.tar.gz libpri-1.2.4 asterisk-core-sounds-en-gsm-1.4.3.tar.gz libpri-1.2-current.tar.gz asterisk-extra-sounds-en-gsm-current.tar.gz zaptel-1.2.12 asterisk-stat-v2_0_1.tar.gz zaptel-1.2.12.tar.gz Any ideas or thoughts on debugging would be appreciated. ___ --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] Is chan_zap.so loaded?
Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded. Also, does autoload in modules.conf take care of it or is it done explicitly? output of lsmod | grep zap: zaptel208388 16 wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w ct4xxp,tor2 crc_ccitt 6465 1 zaptel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any quiet 24 port POE switches out there?
I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. ___ --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] Help needed with Polycom dialplan pattern matching
I'm using Polycom Soundpoint phones and I want to use some extensions beginning with # for features setup. I'm getting the fast busy can't match it signal. I want to match #50 for call forwarding, for instance, and #505551212 to set the call forwarding number and turn it on. I have tftp set up and sip.cfg contains the following: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=#xx.T|[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need advice on dual core processing with *
I have CentOS 4.4 x86_64 running on an Pentium D 830 dual core processor with the smp kernel. Does Asterisk need to be compiled in any special way to gain performance benefits from this setup? ___ --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] Parsing Area Code from CallerID
How would I parse the area code from this variable? Number=2515551212 Sorry for the dense question, I don't seem to be able to find an appropriate function for parsing left to right. ___ --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] Trying to forward calls by using the Callee's context as the forward dial context
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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[asterisk-users] zapata.conf channel variable question
The setvar command below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid=John French 103 mailbox=101 callwaiting=yes threewaycalling=yes transfer=yes channel = 1 setvar=USER=analogPhone ___ --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] Playing a sound file on handset pickup
I've added the ability for a user to record a custom message associated with a special IVR menu for occasions when business will be closed for some non-standard amount of time (Maybe 4 days at Christmas...) They just dial 800, record the message then hang up and dial 801 to enable it. Presumably, when they return after the holiday, they should dial 802 to disable it and return to the normally scheduled menus. But they will most likely forget so I'd like to set up some type of reminder functionality; perhaps playing a message back to them stating that the custom message is still enabled before giving them dialtone or something to the same effect. Is this possible and can anyone offer recommendations? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
