Re: [asterisk-users] Another IP address to block
Where can I find such ip-lists, please? Am 05.06.2012 18:40, schrieb Alejandro Imass: We use complete regional blocks from Wizcraft and blocking at minimum all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We block almost anything that is not our actual customer market and screw the rest. On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavezcur...@telecomabmex.com wrote: Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- _ -- 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 -- _ -- 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] Another IP address to block
On 06-06-12 11:41, Thorsten Göllner wrote: Where can I find such ip-lists, please? http://www.ipdeny.com/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Another IP address to block
Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Another IP address to block
We use complete regional blocks from Wizcraft and blocking at minimum all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We block almost anything that is not our actual customer market and screw the rest. On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavez cur...@telecomabmex.com wrote: Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- _ -- 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] another non-root problem: unable to set utime ??
On Sat, 7 Apr 2012, sean darcy wrote: I'm trying to run asterisk as asterisk. Which is harder than I thought. 10.3.0. When I put a callfile into /var/spool/asterisk/outgoing, I get this warning: Unable to set utime on /var/spool/asterisk/outgoing/callfile.call: Operation not permitted ls -l /var/spool . drwxr-x---. 9 asterisk asterisk 4096 Apr 7 21:41 asterisk ls -l /var/spool/asterisk ... drwxrwx---. 2 asterisk asterisk 4096 Apr 7 21:14 outgoing Do 'ps -U asterisk' or 'ls -l /var/spool/asterisk/outgoing/callfile.call' yield any clues? Also, just in case you're unaware, creating the call file in the /outgoing/ directory is an invitation for a race condition. A 'better practice' is to create the file in a temporary directory on the same device, write to it, close it and 'mv' it. 'mv' is an 'atomic' operation. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Another State Of The Punctuation Mark question - Vonage
There was a flurry of Vonage is going to unlock SIP activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you BYOD pricing* until you actually place the damn order -- or so it appears to my eyes. 727. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ 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] Another State Of The Punctuation Mark question - Vonage
Vonage has a business offering, but they aren't really structured to provide business quality support. I wouldn't use them for a business. For several years now, we've used VoicePulse Connect http://connect.voicepulse.com/ for our Asterisk IAX and SIP trunks. Ravi and KP are both technical guys and know Asterisk extremely well. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Tuesday, September 11, 2007 5:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Another State Of The Punctuation Mark question - Vonage There was a flurry of Vonage is going to unlock SIP activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you BYOD pricing* until you actually place the damn order -- or so it appears to my eyes. 727. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ 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 ___ 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] Another State Of The Punctuation Mark question - Vonage
On Tue, Sep 11, 2007 at 08:56:53AM -0400, Jay R. Ashworth wrote: There was a flurry of Vonage is going to unlock SIP activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you BYOD pricing* until you actually place the damn order -- or so it appears to my eyes. Broadvoice can't handle multiple lines being billed to the same account and using the same SIP credentials, which is probably not too large a deal for a 4 line install, but would quickly become unmanageable for anything larger. jeff 727. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 -- Jeff Bachtel ([EMAIL PROTECTED],TAMU) http://www.cepheid.org/~jeff The sciences, each straining in [finger [EMAIL PROTECTED] for PGP key] its own direction, have hitherto harmed us little; - HPL, TCoC ___ 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] Another State Of The Punctuation Mark question - Vonage
On Tue, Sep 11, 2007 at 09:32:06AM -0700, Eric Chamberlain wrote: For several years now, we've used VoicePulse Connect http://connect.voicepulse.com/ for our Asterisk IAX and SIP trunks. Ravi and KP are both technical guys and know Asterisk extremely well. They'd better be good; their business price is twice everyone elses. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ 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] Another State Of The Punctuation Mark question - Vonage
On Tue, 11 Sep 2007, Jeff Bachtel wrote: Broadvoice can't handle multiple lines being billed to the same account and using the same SIP credentials, which is probably not too large a deal for a 4 line install, but would quickly become unmanageable for anything larger. So it is not easy to provision with them, say, a PRI worth of call appearances off a single SIP contactable? How does one manage this relationship when you need to order large amounts of end-user trunks? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ 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] Another State Of The Punctuation Mark question - Vonage
On Tue, Sep 11, 2007 at 03:53:56PM -0400, Alex Balashov wrote: On Tue, 11 Sep 2007, Jeff Bachtel wrote: Broadvoice can't handle multiple lines being billed to the same account and using the same SIP credentials, which is probably not too large a deal for a 4 line install, but would quickly become unmanageable for anything larger. So it is not easy to provision with them, say, a PRI worth of call appearances off a single SIP contactable? How does one manage this relationship when you need to order large amounts of end-user trunks? Well, it sounds like you go somewhere else. I'm investigating Voicepulse, as someone else suggested. I don't have back CDR to feed them for comparative pricing, so I'm going to have to go disassemble a years worth of Vonage bills. Luckily, I *have* a years worth, right there on line. I don't see TBCT or network-outage forwarding though, in my as yet limited investigation. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ 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
[asterisk-users] Another Faxing Question
This probably came up before, but I have a faxing question for everyone.
I have a simple extension setup to use rxfax to receive faxes sent to
asterisk. It is:
exten = s,1,Answer()
exten = s,n,AbsoluteTimeout(300)
exten =
s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEID}.tif)
exten = s,n,rxfax(${FAXFILE})
exten = s,n,System(/usr/bin/mailfax ${ARG1} ${FAXFILE} ${ARG2})
exten = s,n,Hangup()
exten = T,1,Hangup()
I read that you had to put a AbsoluteTimeout in there, or it might not
hang up. My questions then are... why wouldn't it hang up without the
timeout, and what if the fax really is that large? We sometimes get
faxes over 150 pages.
Rob
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RE: [asterisk-users] Another Faxing Question
In my (limited) experience with rxfax, it has issues with large faxes. I
soon gave up on rxfax and switched to hylafax (which works much better).
Check the wiki for installation instructions. (And hylafax will
correctly hangup when the fax has completed/failed/whatever.)
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Friday, March 09, 2007 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Another Faxing Question
This probably came up before, but I have a faxing question for
everyone.
I have a simple extension setup to use rxfax to receive faxes sent to
asterisk. It is:
exten = s,1,Answer()
exten = s,n,AbsoluteTimeout(300)
exten =
s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEI
D}
.tif)
exten = s,n,rxfax(${FAXFILE})
exten = s,n,System(/usr/bin/mailfax ${ARG1} ${FAXFILE} ${ARG2})
exten = s,n,Hangup()
exten = T,1,Hangup()
I read that you had to put a AbsoluteTimeout in there, or it might not
hang up. My questions then are... why wouldn't it hang up without the
timeout, and what if the fax really is that large? We sometimes get
faxes over 150 pages.
Rob
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[asterisk-users] Another Issue with 1.4
Hi so with my setup of asterisk 1.4 and installing freepbx on it, I have everything working fine now except one thing, the remote console keeps crashing after a reload. Any ideas? ___ --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] Another (quick) Polycom 501 question
Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)?Sort of likewhat a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another (quick) Polycom 501 question
Hi Mike, As far as I know, you need to at least start the dialing (ie New call, speaker, etc) for the digitmap to even come into play. The only settings that I am aware of that you can try to change are dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf. Kevin Mike wrote: Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)? Sort of like what a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] another question about hardware for using with asterisk
On Sun, 7 May 2006, Tofik Suleymanov wrote: Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov I'll pipe in on this one. We got a few D-Link's in to test, and for some strange reason, they're the only ones that won't hold a steady IP address, and haven't been stable at all. You're better off getting cisco's or polycom's, those are designed much better and seem to work really well. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] another question about hardware for using with asterisk
On 7 May 2006, at 16:16, Aaron Daniel wrote: On Sun, 7 May 2006, Tofik Suleymanov wrote: Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov I'll pipe in on this one. We got a few D-Link's in to test, and for some strange reason, they're the only ones that won't hold a steady IP address, and haven't been stable at all. You're better off getting cisco's or polycom's, those are designed much better and seem to work really well. I like the elmeg 290 (cheaper clone of the older snom) It looks, feels and sounds like a 'real' office phone but doesn't dominate a desk. I only have 2, so I can't comment on mass deployment ! I also don't know if they sell outside Europe. Tim. Tim Panton [EMAIL PROTECTED] ___ --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] Another undefined pri_restart failure
Fred Noris wrote: Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL = (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style chan_mgcp.so (0x1) loaded RTLD_LOCAL = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_zap.so]Apr 25 03:36:41 WARNING[8269]: loader.c:718 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart Apr 25 03:36:41 WARNING[8269]: loader.c:850 print_and_load: Loading module chan_zap.so failed! I modified modules.conf to add noload = res_snmp.so, because it fails. I've tried recompiling libpri and everything and modifying path variables. Please help!! It looks like you are using Zaptel/libpri 1.0.x with Asterisk 1.2.x. Don't do that. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Another undefined pri_restart failure
Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL = (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style chan_mgcp.so (0x1) loaded RTLD_LOCAL = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_zap.so]Apr 25 03:36:41 WARNING[8269]: loader.c:718 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart Apr 25 03:36:41 WARNING[8269]: loader.c:850 print_and_load: Loading module chan_zap.so failed! I modified modules.conf to add noload = res_snmp.so, because it fails. I've tried recompiling libpri and everything and modifying path variables. Please help!! ___ For the record, if it is of help Env is: LESSKEY=/etc/lesskey.bin NNTPSERVER=news INFODIR=/usr/local/info:/usr/share/info:/usr/info MANPATH=/usr/share/man:/usr/local/man:/usr/X11R6/man:/opt/gnome/share/man KDE_MULTIHEAD=false SSH_AGENT_PID=6720 HOSTNAME=ScottSuSE DM_CONTROL=/var/run/xdmctl GNOME2_PATH=/usr/local:/opt/gnome:/usr XKEYSYMDB=/usr/X11R6/lib/X11/XKeysymDB GPG_AGENT_INFO=/tmp/gpg-NIZ0pv/S.gpg-agent:17362:1 HOST=ScottSuSE TERM=xterm SHELL=/bin/bash PROFILEREAD=true HISTSIZE=1000 XDM_MANAGED=/var/run/xdmctl/xdmctl-:1,maysd,mayfn,sched,rsvd,method=classic GTK2_RC_FILES=/etc/opt/gnome/gtk-2.0/gtkrc:/opt/gnome/share/themes//Qt/gtk-2.0/gtkrc:/root/.gtkrc-2.0-qtengine:/root/.kde/share/config/gtkrc-2.0 GTK_RC_FILES=/etc/opt/gnome/gtk/gtkrc:/root/.gtkrc:/root/.kde/share/config/gtkrc GNOME_PATH=:/opt/gnome:/usr GS_LIB=/root/.fonts WINDOWID=46137351 OLDPWD=/etc/asterisk QTDIR=/usr/lib/qt3 XSESSION_IS_UP=yes KDE_FULL_SESSION=true GROFF_NO_SGR=yes JRE_HOME=/usr/lib/jvm/java/jre USER=root LS_COLORS=no=00:fi=00:di=01;34:ln=00;36:pi=40;33:so=01;35:do=01;35:bd=40;33;01:cd=40;33;01:or=40;31:ex=00;32:*.cmd=00;32:*.exe=01;32:*.com=01;32:*.bat=01;32:*.btm=01;32:*.dll=01;32:*.tar=00;31:*.tbz=00;31:*.tgz=00;31:*.rpm=00;31:*.deb=00;31:*.arj=00;31:*.taz=00;31:*.lzh=00;31:*.zip=00;31:*.zoo=00;31:*.z=00;31:*.Z=00;31:*.gz=00;31:*.bz2=00;31:*.tb2=00;31:*.tz2=00;31:*.tbz2=00;31:*.avi=01;35:*.bmp=01;35:*.fli=01;35:*.gif=01;35:*.jpg=01;35:*.jpeg=01;35:*.mng=01;35:*.mov=01;35:*.mpg=01;35:*.pcx=01;35:*.pbm=01;35:*.pgm=01;35:*.png=01;35:*.ppm=01;35:*.tga=01;35:*.tif=01;35:*.xbm=01;35:*.xpm=01;35:*.dl=01;35:*.gl=01;35:*.wmv=01;35:*.aiff=00;32:*.au=00;32:*.mid=00;32:*.mp3=00;32:*.ogg=00;32:*.voc=00;32:*.wav=00;32: DESKTOP_LAUNCH=kde-open OPENWINHOME=/usr/openwin XNLSPATH=/usr/X11R6/lib/X11/nls SSH_AUTH_SOCK=/tmp/ssh-IWHyx6676/agent.6676 HOSTTYPE=x86_64 SESSION_MANAGER=local/ScottSuSE:/tmp/.ICE-unix/6784 FROM_HEADER= PAGER=less XDG_CONFIG_DIRS=/usr/local/etc/xdg/:/etc/xdg/:/etc/opt/gnome/xdg/ LD_HWCAP_MASK=0x2000 KONSOLE_DCOP=DCOPRef(konsole-6808,konsole) MINICOM=-c on GNOMEDIR=/opt/gnome DESKTOP_SESSION=default PATH=/sbin:/usr/sbin:/usr/local/sbin:/opt/kde3/sbin:/opt/gnome/sbin:/root/bin:/usr/local/bin:/usr/bin:/usr/X11R6/bin:/bin:/usr/games:/opt/gnome/bin:/opt/kde3/bin:/usr/lib/mit/bin:/usr/lib/mit/sbin CPU=x86_64 JAVA_BINDIR=/usr/lib/jvm/java/bin KONSOLE_DCOP_SESSION=DCOPRef(konsole-6808,session-1) INPUTRC=/etc/inputrc PWD=/usr/src/asterisk/libpri [EMAIL PROTECTED] JAVA_HOME=/usr/lib/jvm/java LANG=POSIX PYTHONSTARTUP=/etc/pythonstart SDK_HOME=/usr/lib/jvm/java SSH_ASKPASS=/usr/lib64/ssh/x11-ssh-askpass TEXINPUTS=::/root/.TeX:/usr/share/doc/.TeX:/usr/doc/.TeX:/root/.TeX:/usr/share/doc/.TeX:/usr/doc/.TeX JDK_HOME=/usr/lib/jvm/java SHLVL=2 HOME=/root LESS_ADVANCED_PREPROCESSOR=no OSTYPE=linux LS_OPTIONS=-a -N --color=tty -T 0 XCURSOR_THEME=crystalwhite WINDOWMANAGER=/usr/bin/dbus-launch --sh-syntax --exit-with-session /usr/X11R6/bin/kde GTK_PATH=/usr/local/lib/gtk-2.0:/opt/gnome/lib/gtk-2.0:/usr/lib/gtk-2.0 LESS=-M -I MACHTYPE=x86_64-suse-linux LOGNAME=root GTK_PATH64=/usr/local/lib64/gtk-2.0:/opt/gnome/lib64/gtk-2.0:/usr/lib64/gtk-2.0 CVS_RSH=ssh XDG_DATA_DIRS=/usr/local/share/:/usr/share/:/etc/opt/kde3/share/:/opt/kde3/share/:/opt/gnome/share/ ACLOCAL_FLAGS=-I /opt/gnome/share/aclocal LC_CTYPE=en_US.UTF-8 DBUS_SESSION_BUS_ADDRESS=unix:abstract=/tmp/dbus-z1RTWWV1Gq,guid=3beb4d44c3081877355afd4083cca800 PKG_CONFIG_PATH=/usr/local/lib/pkgconfig:/usr/local/share/pkgconfig:/usr/lib64/pkgconfig:/usr/share/pkgconfig:/opt/kde3/lib64/pkgconfig:/opt/gnome/lib64/pkgconfig:/opt/gnome/lib64/pkgconfig:/opt/gnome/share/pkgconfig LESSOPEN=lessopen.sh %s USE_FAM= INFOPATH=/usr/local/info:/usr/share/info:/usr/info:/opt/gnome/share/info DISPLAY=:1 XAUTHLOCALHOSTNAME=ScottSuSE LESSCLOSE=lessclose.sh %s
[Asterisk-Users] another nat question
Any disadvantage to always setting nat=yes for all UAs just in case they end up behind a NAT at some point? Canreinvite=no is always set since a few of our features require it (transfers, etc.) What is the impact of qualify=yes for 250-500 UAs? ___ --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] Another cisco question
Sorry about the unrelated questions about cisco phones, but does anyone know how to set the second line up as a speed dial in the config file? Or is that specifically a per-user basis setting? Aaron ___ --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] Another problem on queues
Hello
all,
I have
been posting some questions about this problems that I cannot yet solve, but I
think I have a better diagostic, so maybe someone can give me a clue why it is
happenning.
I have
Asterisk + AMPconfigured as a PBX with a Customer Center Queue with 4
agents that login/logout dinamically.
If
there are no agents, queue timesout and gets derived to another queue that
somebody answers as last resort or waits there.
If
there is at least one agent logged in, but it is busy, dialparties.agi detects
that that extension has no callwaiting, no callforward, no voicemail, and hangs
up the call inmediately with a "nobody is available to take your call right now"
message, making the queue useless.
My
PSTN connection is an AS5300 in SIP, my extensions are analog phones connected
to an Audiocodes MP108-FXS with SIP.
This
is the output from CLI with High Verbosity:
XXX.XXX.XXX.XXX is the IP of the AS5300, 8521 and 8522 are the only two
agents in the queue that have inbound calls in progress when a third call
arrives and this happens. 8500 is the queue number
-- Executing SetVar("SIP/XXX.XXX.XXX.XXX-43921110", "FROM_DID=1154538500") in
new stack -- Executing
Goto("SIP/XXX.XXX.XXX.XXX-43921110", "ext-did|1154538500|1") in new
stack -- Goto (ext-did,1154538500,1)
-- Executing Goto("SIP/XXX.XXX.XXX.XXX-43921110", "ext-queues|8500|1") in new
stack -- Goto (ext-queues,8500,1) --
Executing Answer("SIP/XXX.XXX.XXX.XXX-43921110", "") in new
stack -- Executing
SetCIDName("SIP/XXX.XXX.XXX.XXX-43921110", "XXX.XXX.XXX.XXX") in new
stack -- Executing SetVar("SIP/XXX.XXX.XXX.XXX-43921110",
"MONITOR_FILENAME=/var/spool/asterisk/monitor/q") in new
stack -- Executing Queue("SIP/XXX.XXX.XXX.XXX-43921110",
"8500|t|||300") in new stack -- Started music on hold,
class 'operadores', on SIP/XXX.XXX.XXX.XXX-43921110 --
Executing Macro("Local/[EMAIL PROTECTED],2",
"exten-vm|[EMAIL PROTECTED]|8521") in
new stack -- Executing SetVar("Local/[EMAIL PROTECTED],2",
"FROMCONTEXT=exten-vm") in new stack -- Executing
Macro("Local/[EMAIL PROTECTED],2",
"record-enable|8521|IN") in new stack -- Executing
GotoIf("Local/[EMAIL PROTECTED],2",
"0 0?2:4") in new stack -- Goto
(macro-record-enable,s,4) -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?5:8") in new stack -- Goto
(macro-record-enable,s,8) -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?9:12") in new stack -- Goto
(macro-record-enable,s,12) -- Executing DBget("Local/[EMAIL PROTECTED],2",
"RecEnable=RECORD-IN/8521") in new stack -- DBget:
varname=RecEnable, family=RECORD-IN, key=8521 -- DBget:
Value not found in database. -- Executing SetVar("Local/[EMAIL PROTECTED],2",
"CALLFILENAME=20050805-43-1123251103.2060") in new
stack -- Called Local/[EMAIL PROTECTED]
-- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?15:99") in new stack -- Goto
(macro-record-enable,s,99) -- Executing NoOp("Local/[EMAIL PROTECTED],2",
"NO RECORDING NEEDED") in new stack -- Executing
GotoIf("Local/[EMAIL PROTECTED],2",
"1?novm|1:4") in new stack -- Goto
(macro-exten-vm,novm,1) -- Executing Macro("Local/[EMAIL PROTECTED],2",
"dial|120|tr|8521") in new stack -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?4:2") in new stack -- Goto
(macro-dial,s,2) -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?4:3") in new stack -- Goto
(macro-dial,s,3) -- Executing SetCIDName("Local/[EMAIL PROTECTED],2",
"XXX.XXX.XXX.XXX") in new stack -- Executing AGI("Local/[EMAIL PROTECTED],2",
"dialparties.agi") in new stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi --
dialparties.agi: request = dialparties.agi --
dialparties.agi: priority = 4 -- dialparties.agi:
extension = s -- dialparties.agi: language =
en -- dialparties.agi: accountcode
= -- dialparties.agi: uniqueid =
1123251103.2060 -- dialparties.agi: channel = Local/[EMAIL PROTECTED],2
-- dialparties.agi: callerid =
XXX.XXX.XXX.XX.XXX.XXX.XXX --
dialparties.agi: context = macro-dial --
dialparties.agi: type = Local -- dialparties.agi:
rdnis = unknown -- dialparties.agi: enhanced =
0.0 -- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID is not set --
dialparties.agi: Added extension 8521 to extension map
-- dialparties.agi: Extension 8521 cf is disabled
-- dialparties.agi: Extension 8521 do not disturb is disabled ==
Parsing '/etc/asterisk/manager.conf': Found == Parsing
'/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged
on from 127.0.0.1 == Manager 'admin' logged off from
127.0.0.1 dialparties.agi: Extension 8521 has call waiting
disabled dialparties.agi: Max calls of 1 exceeded - deleting from
dial dialparties.agi: Dial string is empty - nothing to do
dialparties.agi: Was direct call, jumping to priority 26
-- AGI Script Executing Application: (NoOp) Options: () --
AGI Script dialparties.agi completed, returning 0 --
Executing Wait("Local/[EMAIL PROTECTED],2",
"1") in new stack -- Executing
Re: [Asterisk-Users] (Another) Queue log analyser
hi is this stuff still available? roy On 14. okt. 2004, at 16.10, Ben Merrills wrote: I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another OH323 Problem
Title: Message anyone got any ideas on this? TDM H323 Gateway SIP Inbound H.323 call 'ip$200.93.237.82:12984/2853' detected.Channel OH323/R2853 created and attached for inbound H.323 call 'ip$200.93.237.82:12984/2853'.Setting channel 'OH323/R2853' (ip$200.93.237.82:12984/2853) native format to g723!Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 1.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 2.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 3.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 4.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 5.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 6.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 7.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 8.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 9.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count is 10.Call 'ip$200.93.237.82:12984/2853' cleared.Call 'ip$200.93.237.82:12984/2853' without owner has already been cleared (2). any and all help would be appreciated... jeromy Global reach, local touch... Jeromy GrimmettCEO Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, LA 71301 [EMAIL PROTECTED]IM: MSN: [EMAIL PROTECTED]http://www.comuniquemonos.com tel: fax: mobile: +593 (4) 287 3854(501) 646-0680+593 (9) 366 6521 Add me to your address book... Want a signature like this? ___ 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] (another) cisco 7960 question
My 7960 is configured for two lines, and I can turn the other appearance buttons into speed dials from the menus, but is there any way to program the speed dials in the SIPmacaddress.conf file? /edg ___ 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] (another) cisco 7960 question
My 7960 is configured for two lines, and I can turn the other appearance buttons into speed dials from the menus, but is there any way to program the speed dials in the SIPmacaddress.conf file? You can not: http://tinyurl.com/az4fp -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ 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] (another) cisco 7960 question
Use the Directory or Services to create a speed dial list. On 5/22/05, Nabeel Jafferali [EMAIL PROTECTED] wrote: My 7960 is configured for two lines, and I can turn the other appearance buttons into speed dials from the menus, but is there any way to program the speed dials in the SIPmacaddress.conf file? You can not: http://tinyurl.com/az4fp -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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] another voipjet question
Haven't done this yet Art but I will try it today at the
office...Thanks Jonathan
On Mon, 28 Mar 2005 00:30:32 -0600, Tim Litwiller [EMAIL PROTECTED] wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I want asterisk to use my pots line for local calls and voipjet for long
distance after the initial 100 free minutes my line provider give with
our plan. but to failover if one is busy and the other isn't.
Art Zemon wrote:
Jon Walsh wrote:
No Dice so far, anyone now how to add anIAX trunk? What are the
settings exactly?
Jon,
It took me awhile to get voipjet working with AAH because I was stubborn
and wanted to get it going through the AMP interface, instead of by hand
crafting the .conf files.
The trick was that I had to make *two* trunks for voipjet. The second
trick was to ignore a buglette in AMP. Here is what I did:
1. Create an IAX trunk. You *must* enter trunk name [EMAIL PROTECTED]
where 1234 is your voipjet ID. Cut 'n' paste all of the other
details from voipjet's site into the outgoing peer details window.
Leave all of the incoming stuff and the registration string blank;
you can't receive calls through voipjet.
2. Create a second IAX trunk. You *must* name this trunk voipjet. I
entered all of the same info here, too, but I think that all you
need in the peer details is the host= line.
If you don't create the second trunk, you will get a message in the log
that is something like voipjet: host not found when * tries to dial
with the string IAX2/[EMAIL PROTECTED]/16365551212
The buglette is that if you try to re-edit the first trunk, the first
digit from the trunk name will be missing. Fear not, the trunk name is
stored correctly THE FIRST TIME YOU SAVE. After that, AMP will mess it
up and you will need to remember to manually correct the trunk name if
you edit and save.
Cheers,
-- Art Z.
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Re: [Asterisk-Users] another voipjet question
I'm working on it - I only started a week ago - and then I didn't know I
wanted to do all these other things with it. * is adictive!
Art Zemon wrote:
Tim Litwiller wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Tim,
I did not use those lines. If you set up the two trunks as I described,
AAH will route calls out through voipjet. You don't have to manually add
those lines.
I want asterisk to use my pots line for local calls and voipjet for
long distance after the initial 100 free minutes my line provider give
with our plan. but to failover if one is busy and the other isn't.
Ahhh... *now* I think you need to get familiar with writing Asterisk
config files. :-)
-- Art Z.
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[Asterisk-Users] another voipjet question
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I want asterisk to use my pots line for local calls and voipjet for long
distance after the initial 100 free minutes my line provider give with
our plan. but to failover if one is busy and the other isn't.
Art Zemon wrote:
Jon Walsh wrote:
No Dice so far, anyone now how to add anIAX trunk? What are the
settings exactly?
Jon,
It took me awhile to get voipjet working with AAH because I was stubborn
and wanted to get it going through the AMP interface, instead of by hand
crafting the .conf files.
The trick was that I had to make *two* trunks for voipjet. The second
trick was to ignore a buglette in AMP. Here is what I did:
1. Create an IAX trunk. You *must* enter trunk name [EMAIL PROTECTED]
where 1234 is your voipjet ID. Cut 'n' paste all of the other
details from voipjet's site into the outgoing peer details window.
Leave all of the incoming stuff and the registration string blank;
you can't receive calls through voipjet.
2. Create a second IAX trunk. You *must* name this trunk voipjet. I
entered all of the same info here, too, but I think that all you
need in the peer details is the host= line.
If you don't create the second trunk, you will get a message in the log
that is something like voipjet: host not found when * tries to dial
with the string IAX2/[EMAIL PROTECTED]/16365551212
The buglette is that if you try to re-edit the first trunk, the first
digit from the trunk name will be missing. Fear not, the trunk name is
stored correctly THE FIRST TIME YOU SAVE. After that, AMP will mess it
up and you will need to remember to manually correct the trunk name if
you edit and save.
Cheers,
-- Art Z.
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Re: [Asterisk-Users] another voipjet question
Tim Litwiller wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Tim,
I did not use those lines. If you set up the two trunks as I described,
AAH will route calls out through voipjet. You don't have to manually add
those lines.
I want asterisk to use my pots line for local calls and voipjet for
long distance after the initial 100 free minutes my line provider give
with our plan. but to failover if one is busy and the other isn't.
Ahhh... *now* I think you need to get familiar with writing Asterisk
config files. :-)
-- Art Z.
--
Art Zemon, President
Hen's Teeth Network http://www.hens-teeth.net/
Voice Fax: (866)HENS-NET or (636)447-3030
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meaningful subject [was: Re: [Asterisk-Users] Another Newbie Question]
On Wed, Mar 09, 2005 at 06:38:24PM +1100, Callum McGillivray wrote: Hey all, Hi, welcome to this list My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? I figure it's possible. As I said, probably blindingly obvious. but my techies have gone home for the evening and I was looking for an answer before I left. Suppose someone will have the same question a year from now. He'll try to do the Right Thing and search the archives of this list first. He may get some hits for his search from this thread, but will dismiss them, because the title of the thread was a newbie question and gives no hint to the fact that we're talking about connecting extensions. Cheers -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] Another Newbie Question
Hey all, My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? As I said, probably blindingly obvious but my techies have gone home for the evening and I was looking for an answer before I left. Thanks, Callum ___ 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] Another Newbie Question
Callum McGillivray wrote: Hey all, My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? As I said, probably blindingly obvious but my techies have gone home for the evening and I was looking for an answer before I left. You could do that with two tin cans and a string! ;-P In all seriousness, the answer to your question is: yes, Asterisk can do that, and a whole lot more. Cheers, -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.4 - Release Date: 07/03/2005 ___ 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] (another try) Dialing phone number and extension together to avoid listening to voice menu (incoming call)
Hello,
Sorry for reposting the message, but I'm not sure the first post went
through.
I'm trying to figure out how to get Asterisk to dial an extension when a
call comes from the outside and contains the extension already.
(Somebody wants to call a user of Asterisk with extension 111 from the
outside)
For example: I've hooked Asterisk to sipgate.de and received a landline
phone number (say 0781205237).
Now if you dial 0781205237 and and an extension altogether
(0781205237111) I would like Asterisk to redirect the call to the
extension 111, without having to listen to the greetings message and
then typing the extension on the keypad.
Please help me to figure it out. Any suggestions and code excerpts would
be highly appreciated.
Also, I was trying to use a voice menu setup for that, so that when the
user dials 0781205237, he/she would listen to the greeting and then
can enter the extension on the phone. However, I couldn't get this to
work either.
Here is the excerpt of my extensions.conf:
***
; defining the voice menu for incoming calls:
[fhostaffmenu]
exten = s,1,Ringing; Make them comfortable with
some seconds of ringback
exten = s,2,Answer ; Answer the line
exten = s,3,DigitTimeout(1); Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout(3) ; Set Response Timeout to 10 seconds
exten = s,5,Read(mynumber,beep,3) ; Read DTMF input and save it into
mynumber variable exten =
s,6,Macro(fhostaff,${mynumber},SIP/${mynumber}) ; dial the extension
that is saved in mynumber
***
When I execute this, it says that User entered ''. Why wouldn't it
read the numbers punched on the phone? The Voicemail works very well.
I use dtmfmode = rfc2833 and iLBC codec.
Also, please check if the comments I made to the code below are correct.
Thank you very much,
Roman Zhovtulya
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Re: [Asterisk-Users] (another try) Dialing phone number and extension together to avoid listening to voice menu (incoming call)
***
; defining the voice menu for incoming calls:
[fhostaffmenu]
exten = s,1,Ringing ; Make them comfortable with
some seconds of ringback
exten = s,2,Answer ; Answer the line
You haven't actually given them any ringing, you need to add this:
exten = s,3,wait(2) ; Give them 2 seconds of ringing
exten = s,4,DigitTimeout(1) ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout(3) ; Set Response Timeout to 10 seconds
Rather than doing the below, if you simply stop all processing at this
point, and don't have any more extensions, then asterisk will wait 3
seconds for the user to press a number, then 1 second for each extra
number. When they don't press a number for more than the 1 second, or
asterisk matches an extension, then it will try to dial the entered
number.
exten = s,5,Read(mynumber,beep,3) ; Read DTMF input and save it into
mynumber variable exten =
s,6,Macro(fhostaff,${mynumber},SIP/${mynumber}) ; dial the extension
that is saved in mynumber
***
OK, hard to get asterisk to do this, but something like:
exten = _XXX.,Macro(fhostaff,${mynumber},SIP/${mynumber})
So, the user can dial 3 or more digits, and then it will go to your
macro.
You can also add:
exten =
i,1,playback(some_file_to_say_you_entered_an_invalid_extension_try_again)
exten = i,2,Goto(s,4)
and also:
exten =
t,1,playback(some_file_to_say_you_did_not_enter_an_extension_try_again)
exten = t,2,Goto(s,4)
I hope that helps you...
Regards,
Adam
--
--
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Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
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RE: [Asterisk-Users] Another BroadVoice Problem
What ports do I need to have opened? I have completely opened up the asterisk server for UDP. Router | Switch And asterisk and all other phones are connected to the switch.. The router has an ACL with the asterisk server being allowed all UDP. -Original Message- From: John Sawa [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 26, 2005 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Another BroadVoice Problem This is a firewall/NAT issue. The UDP packets on your inbound RTP stream and being dropped somewhere along the line, most likely at your firewall/router or your SIP messages contain a non routable address so BroadVoice is sending your RTP stream to a bad destination. You will need to include your SIP messages and a network topology if you would like someone on the list to chime in. -John Manjit Riat wrote: I finally got my incoming and outgoing working but outgoing I cannot hear the called person, but the called person can hear me. On incoming everything works perfect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another BroadVoice Problem
I finally got my incoming and outgoing working but outgoing I cannot hear the called person, but the called person can hear me. On incoming everything works perfect. ___ 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] Another Asterisk Certification - Ordained Ministers, oh my!
Awesome now i'm a minister! On Wed, 22 Dec 2004 12:45:51 -0600, Kristian Kielhofner [EMAIL PROTECTED] wrote: Alexander Lopez wrote: Agreed, You have a strong point about the Monopoly aspect of the whole thing. My .02 would be to have this be a Digium product. Heck, Mark DID invent the thing and HE holds the copyright to it. I have faith in Mark and what he can do when he gets back from France. When I taught Sun and SCO classes (flame throwers down, please) it was SCO corporate that provided the Certification, albeit outsourced but it was from the same people that put out the OS. In order to be an instructor you needed to have taken the course or have 'tested out' and had to have a score above 90 on your exam. (70 was passing). This allows the certification program to grow as your instructors know the course material and will be able to teach students with real world situations. You would purchase the course materials at a 60% discount. For the 40% they would handle certs, and testing at Sylvan. I think that it is pretty obvious that this is a joke, aside from the striking similarity to internet minister, the payment buttons go no where, they POINT OUT that the funds go to the Caymen's, etc. Check this out: http://www.ulc.org/ Most of my friends have ordained themselves after some long nights of partying or whatnot. The site claims to have ordained over 400,000 people. If I am not mistaken you even get a certificate that you can print out so you can park in clergy spots at hospitals, etc... Let's get back to Asterisk chatter! (After you all ordain yourselves, of course)! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Asterisk Certification
Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ 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] Another Asterisk Certification
How much time did you waste on that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 10:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
I'm sure it took several hours, but, hey, he only has to sell one to get his money back (: Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 22 Dec 2004, Luke Catranis wrote: How much time did you waste on that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 10:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-22%5C3a0f4f41805f4aa297eb4dbf29c2b394C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Another Asterisk Certification
The place it has for me is that my work is hesitant to pay for additional training if there's nothing to show for it, like a certification. They can't tell if I go to a $2000 training course and just goof off, there's no tangible goods that they can see. If I got them to shell out $3000 for my training, and I get the certification, then the powers that be will feel better about themselves and justified in spending the money. As far as I'm concerned, if it convinces them to pay to make me a better/smarter person, then I'm all for it. However, if I had to pay for it myself, I'd only consider it if it was truly worth it. I don't want to waste hours on how to setup and install linux and how to create user accounts in sip.conf and iax.conf and how to make an extension call a phone, I can do that in my sleep, I want to learn about the most advanced features in Asterisk, some of those little commands that people tend to forget about in the day to day configuration. And I would want to learn some very neat tips and tricks and such. If it offered that, then I'd scrounge up the $3000. Especially if it lands me a $50,000/yr+ Job. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 10:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. Yeah, it worked wonders for Microsoft. ___ 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] Another Asterisk Certification
Once you've been ordained, do you have to wear black robes and a white collar while working on Asterisk? :-) Steve Brian West wrote: No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Asterisk Certification
But wait, that's not all! I, too, have a laser printer! If you send me $50, I'll fire you off a certificate too! You can be a Certified Asterisk Certification Certificate Buyer! Enough. It is what it is. Don't like it? Don't pay for it. Think it's a joke? Sure, but it's the same sick joke visited upon us time and time again by the illustrious HR and Vendor Services departments of corporate America. Let's take this to another list (perhaps [EMAIL PROTECTED] :) ) and get on with fixing our little Asterisk problems. Clint On Dec 22, 2004, at 10:53 AM, Brian West wrote: No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Asterisk Certification
Hello Guys, I think this is not bad (Certification) While is a real certification like Cisco , Novell, etc. how many of us have a cisco certification or even Micro-$hit. In my point of vew this 3K buck are well spended if I want to have the skills quick to put hand-on. and as per Brian comment , not just a simple config. just think about how much you guys charge per hour, and how many hours you all ready spend in learning, those 3000 bucks are really a joke. I think those certified guys need to place the training agenda in the Commercial posrt. I really like tho se certifications (Y) Regards HA On Wed, 22 Dec 2004 09:53:47 -0600, Brian West [EMAIL PROTECTED] wrote: No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. What makes me different than the person next door selling the same solution. We both have access to the hardware, software and newsgroups. What is there to tell the customer that _I_ am better suited to solve the customers' problem or implement a solution that will work?? Nothing! We are faced with a growing install base and many who put together systems are doing it for the first time. Sure any customers will get fed up with Asterisk and vie never to use it again. Others, like many customers' that I have found like it but need a someone with expertise to help them through the tough configs. Will Asterisk certification help me, depends. My strength not only lie in the configuration and modification of Asterisk but in the whole telecom and networking expeiriance that I have. This is new stuff, you must know about MTU, E1, T1, PRI, FXO, FXS, TCP, UDP, Linux, Iptables, http, Nat, etc. Tell me why you can still talk to someone clearly, but are not able to use a modem past 2400 baud at a hotel, and you'll understand codecs and how they interoperate. We conserned ourselves with being 'left out' of a certification that may not actually reflect what we know is needed to do what we do. I have never or probably never hired someing because of certifications alone. You got um, good! Go fix the problem, that was not covered in 'the book'!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Voip Business Sent: Wednesday, December 22, 2004 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Another Asterisk Certification Hello Guys, I think this is not bad (Certification) While is a real certification like Cisco , Novell, etc. how many of us have a cisco certification or even Micro-$hit. In my point of vew this 3K buck are well spended if I want to have the skills quick to put hand-on. and as per Brian comment , not just a simple config. just think about how much you guys charge per hour, and how many hours you all ready spend in learning, those 3000 bucks are really a joke. I think those certified guys need to place the training agenda in the Commercial posrt. I really like tho se certifications (Y) Regards HA On Wed, 22 Dec 2004 09:53:47 -0600, Brian West [EMAIL PROTECTED] wrote: No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Another Asterisk Certification
certify this ___ 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] Another Asterisk Certification
What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. I never said this was a bad idea... And I'm not against it at all if people wish to pay for it that's fine. I don't however feel that Steve and the boys should have a monopoly on it. From what I was told you'll have to go to Steve and the boys to buy a franchise to become a trainer and pay ... and you think $3,275 USD was a lot... just think what they would charge to become a dCap trainer?!?!?!? That's one problem with this... ANY company should be able to signup to become dCap trainers. The way the whole thing was presented was what the problem was... things were left out and over looked that's what I had the most problem with. Now to be honest if you were going to pay the 3k for the course you should walk out with a something like a Dell 420SC and a T1 card for that price. I feel that would be the only way to really make it balance at that price. bkw ___ 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] Another Asterisk Certification
Man, this is sick! :-))) Isn't there a law against unclearly-marked jokes (except on April 1st, of course)? Some people could even take you seriously! :-)) Most relevant points in the web page: Starting a telephone company or consulting business is easy. We authorize you to perform all Asterisk services including the rites of Interconnection and Arbitration. You will also be authorized to put the “*” symbol after your name – much like: “PHD, MD, DDS, CLU, ETC.” Just picture your business card: John Smith * this in itself gives you the chance to earn extra money to support your Asterisk habit and family Save 50% send CASH or Postal Money Order to our office in the Cayman Islands I'm rolling on the floor here :- Regards, Telmo. On Sun Aug 22 7:24 , 'James Taylor' [EMAIL PROTECTED] sent: Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ 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 SeeqMail - the only email you'll ever need Starts Here Sign up for FREE personalized email today: http://www.seeqmail.com http://www.Grassroots.org/ - Make Change! ___ 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] Another Asterisk Certification? -- This time we might just Unionize
I, being one of the original Microsoft Certified guys, back then you sent them $150 and you got the certificate and some logos (think 1980's certification.) In 1996 I was told by the company I was working for the certification was needed if I was to keep my current salary. What I saw was morons who had time to study for the tests, because they could not code anyway, pass the test and get certified and allowed to keep their pay grade. I took two tests, got my pin and left for a better job with more money. Certification works to show that the person can pass a test, not that they can do anything in the real world. Corporate America likes certifications as it provides them with a comfort level and a box that can be checked off. I have never hired a person based on being certified, I can tell if they can do the job or not by interviewing them. Real programmers just get things done anyway. Would you really want to work at a place that required any kind of certification? I agree that Digium class for $3000 are a good thing, they help people come up to speed and help stop the confusion in the beginning. If customer are asking for it then they should be allowed to buy it. Certification stinks in the real world. In the world of corporations it is necessary evil and part of the game when you sell to the corporations. Lets unionize all the programmers in the world so we can set rates and standards and get the respect we deserve. Oh yeah, I forgot, Unionizing programmers won't work for the same reason women make only 70% of the wages men do. A woman will always undercut another woman to get the job instead teaming together and holding out for better wages. As the subcontinent programmers under cut the US programmers, but in the end you get what you pay for in America. Race The Tyrant Vanderdecken In the Land of the Blind, the One-eyed man is Elvis ___ 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] Another Asterisk Certification
[EMAIL PROTECTED] wrote: Man, this is sick! :-))) Isn't there a law against unclearly-marked jokes (except on April 1st, of course)? Some people could even take you seriously! :-)) I'm rolling on the floor here :- Regards, Telmo. Then I guess you haven't seen this one: http://www.j-walk.com/other/conf/ Steve ___ 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] Another Asterisk Certification
I agree.. No certs needed. I know * better than probably all of your students combined dude.. I agree with BKW.. .o---o. Brian Fertig Network Engineer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 22, 2004 10:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Another Asterisk Certification No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!! This is a joke right? I has to be. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 9:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
Agreed, You have a strong point about the Monopoly aspect of the whole thing. My .02 would be to have this be a Digium product. Heck, Mark DID invent the thing and HE holds the copyright to it. I have faith in Mark and what he can do when he gets back from France. When I taught Sun and SCO classes (flame throwers down, please) it was SCO corporate that provided the Certification, albeit outsourced but it was from the same people that put out the OS. In order to be an instructor you needed to have taken the course or have 'tested out' and had to have a score above 90 on your exam. (70 was passing). This allows the certification program to grow as your instructors know the course material and will be able to teach students with real world situations. You would purchase the course materials at a 60% discount. For the 40% they would handle certs, and testing at Sylvan. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 22, 2004 12:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Another Asterisk Certification What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. I never said this was a bad idea... And I'm not against it at all if people wish to pay for it that's fine. I don't however feel that Steve and the boys should have a monopoly on it. From what I was told you'll have to go to Steve and the boys to buy a franchise to become a trainer and pay ... and you think $3,275 USD was a lot... just think what they would charge to become a dCap trainer?!?!?!? That's one problem with this... ANY company should be able to signup to become dCap trainers. The way the whole thing was presented was what the problem was... things were left out and over looked that's what I had the most problem with. Now to be honest if you were going to pay the 3k for the course you should walk out with a something like a Dell 420SC and a T1 card for that price. I feel that would be the only way to really make it balance at that price. bkw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Asterisk Certification - Ordained Ministers, oh my!
Alexander Lopez wrote: Agreed, You have a strong point about the Monopoly aspect of the whole thing. My .02 would be to have this be a Digium product. Heck, Mark DID invent the thing and HE holds the copyright to it. I have faith in Mark and what he can do when he gets back from France. When I taught Sun and SCO classes (flame throwers down, please) it was SCO corporate that provided the Certification, albeit outsourced but it was from the same people that put out the OS. In order to be an instructor you needed to have taken the course or have 'tested out' and had to have a score above 90 on your exam. (70 was passing). This allows the certification program to grow as your instructors know the course material and will be able to teach students with real world situations. You would purchase the course materials at a 60% discount. For the 40% they would handle certs, and testing at Sylvan. I think that it is pretty obvious that this is a joke, aside from the striking similarity to internet minister, the payment buttons go no where, they POINT OUT that the funds go to the Caymen's, etc. Check this out: http://www.ulc.org/ Most of my friends have ordained themselves after some long nights of partying or whatnot. The site claims to have ordained over 400,000 people. If I am not mistaken you even get a certificate that you can print out so you can park in clergy spots at hospitals, etc... Let's get back to Asterisk chatter! (After you all ordain yourselves, of course)! -- Kristian Kielhofner ___ 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] Another Asterisk Certification
Guys I think I DONT GET THE POINT ,, or all of us didnt get the point. there is too much diference between Certified training and learning course of asterisk. and Course to be certified in asterisk. yes I also know order of factors do not afect the result :) to the guys of metrotel,, to avoid this kinda topics can you place a Index or agenda , so all of usundestandwhat you really mean. regards HA On Wed, 22 Dec 2004 13:17:50 -0500, Alexander Lopez [EMAIL PROTECTED] wrote: Agreed, You have a strong point about the Monopoly aspect of the whole thing. My .02 would be to have this be a Digium product. Heck, Mark DID invent the thing and HE holds the copyright to it. I have faith in Mark and what he can do when he gets back from France. When I taught Sun and SCO classes (flame throwers down, please) it was SCO corporate that provided the Certification, albeit outsourced but it was from the same people that put out the OS. In order to be an instructor you needed to have taken the course or have 'tested out' and had to have a score above 90 on your exam. (70 was passing). This allows the certification program to grow as your instructors know the course material and will be able to teach students with real world situations. You would purchase the course materials at a 60% discount. For the 40% they would handle certs, and testing at Sylvan. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 22, 2004 12:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Another Asterisk Certification What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. I never said this was a bad idea... And I'm not against it at all if people wish to pay for it that's fine. I don't however feel that Steve and the boys should have a monopoly on it. From what I was told you'll have to go to Steve and the boys to buy a franchise to become a trainer and pay ... and you think $3,275 USD was a lot... just think what they would charge to become a dCap trainer?!?!?!? That's one problem with this... ANY company should be able to signup to become dCap trainers. The way the whole thing was presented was what the problem was... things were left out and over looked that's what I had the most problem with. Now to be honest if you were going to pay the 3k for the course you should walk out with a something like a Dell 420SC and a T1 card for that price. I feel that would be the only way to really make it balance at that price. bkw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification (couldn't be a bad thing)
Well give oej and steve some time here ... the project sure couldn't hurt from more enterprise funding... lets just hope some of that makes it way back to the root of the project. Also I was quick to judge their intentions and I shouldn't have been... so guys lets give them some support and see what happens. Oej and Steve I'm sorry for the rush judgment. bkw ___ 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] Another Unable to create channel of type 'Zap' (cause 0) error
.. and from a newbie no less :-) I have configured my BT101, and hooked it up to my * box. All is well. I have entered the following in externsions.conf, and this bit works: exten = 613,1,Answer exten = 613,2,Playback(demo-echotest) exten = 613,3,Echo exten = 613,4,Hangup If I pick up the BT101, and dial 613, sure enough I get the echo test.. All good. I have a TDM400 Card with a single FXO port on it. ztcfg -vv recognises the card as FXS Device (I think that's right though...??) I want to know how to get, say extension 1000 to dial a number on the FXO card.. ie: exten = 1000,1,Answer exten = 1000,2,Dial(Zap/1:555-1234,20,tr) exten = 1000,3,Hangup That should work, shouldn't it? Well it doesn't :-) Hence the error in the subject of this message!.. I'm a total noob, but once I get my head around this, I'm sure I'll have no problems.. Oh, and what extension do I use to reference an incoming call on my FXO port? exten = 1 ?? Alan ___ 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] Another Unable to create channel of type 'Zap' (cause 0) error
On Mon, 2004-12-06 at 17:40, Alan Ingleby wrote: exten = 1000,2,Dial(Zap/1:555-1234,20,tr) Change this to exten = 1000,2,Dial(Zap/1/5551234,20,tr) Oh, and what extension do I use to reference an incoming call on my FXO port? exten = 1 ?? You want the s extension. http://www.voip-info.org/wiki-Asterisk+s+extension -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] (Another) Queue log analyser
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Thursday, October 14, 2004 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] (Another) Queue log analyser I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? Ben, have you gotten any further with this? Would you share your code? ___ 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] (Another) Queue log analyser
Title: (Another) Queue log analyser Ben, I would definitely have use for this application, fantastic start. When will you be making the source available? In my reports I use the CLID to look at calls for different agents i.e. call volume by agent. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney Message: 4 Date: Fri, 15 Oct 2004 09:33:26 +0100 From: Ben Merrills [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] (Another) Queue log analyser To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi there, Cheers for your suggestions, would be great to see the output of some other reports. Logins and logouts are available within the engine, just need to represent them in some way now. What do you suggest would be a good format? Typical duration of login? Only problem might be where someone hasn't logged out before their next login statement (no one here ever logs out, because they're all to slack :) Anything you can send me over would be much appreciated, I have no problems in giving you a pre-release copy so you can give some feedback too. Regards, Ben Merrills Griffin Internet T: 0870 8040862 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wayne Sheppard Sent: 14 October 2004 19:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (Another) Queue log analyser Very nice work Ben, thanks. Here are some additional thoughts - One segmentation that might be useful would be to add outbound calling activities as a either a separate column or even view. On agent stats, it would be useful to see login/logout stamps, login time, ready/not ready time (if this can be tracked, not sure). If you would like, I can send you some example reports that are used in a typical call center, contact me directly if you would find that helpful. Cheers, Wayne Ben Merrills wrote: I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ 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] (Another) Queue log analyser
Hi there, Cheers for your suggestions, would be great to see the output of some other reports. Logins and logouts are available within the engine, just need to represent them in some way now. What do you suggest would be a good format? Typical duration of login? Only problem might be where someone hasn't logged out before their next login statement (no one here ever logs out, because they're all to slack :) Anything you can send me over would be much appreciated, I have no problems in giving you a pre-release copy so you can give some feedback too. Regards, Ben Merrills Griffin Internet T: 0870 8040862 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sheppard Sent: 14 October 2004 19:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (Another) Queue log analyser Very nice work Ben, thanks. Here are some additional thoughts - One segmentation that might be useful would be to add outbound calling activities as a either a separate column or even view. On agent stats, it would be useful to see login/logout stamps, login time, ready/not ready time (if this can be tracked, not sure). If you would like, I can send you some example reports that are used in a typical call center, contact me directly if you would find that helpful. Cheers, Wayne Ben Merrills wrote: I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Another) Queue log analyser
I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ 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] (Another) Queue log analyser
Wow! That\'s great! Our company could really benefit from this level of analysis. Previously we were using Nortel Merridian, and everyone is used to that level of reporting. Your report(s) are the closest I\'ve seen in their ability to provide the necessary statistics to manage a call center. If you have a version ready for testing, please let me know (on or off list) and I\'ll get it installed here. Thank you! Joe -Original Message- From: Ben Merrills Sent: Thursday, 14. Oct 2004 9:10 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] (Another) Queue log analyser I\'ve been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Another) Queue log analyser
I would like the source too so I can re-write it in non-.NET. Probably C or PHP. Matthew - Original Message - From: Joe Dennick [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 14, 2004 11:57 AM Subject: Re: [Asterisk-Users] (Another) Queue log analyser Wow! That\'s great! Our company could really benefit from this level of analysis. Previously we were using Nortel Merridian, and everyone is used to that level of reporting. Your report(s) are the closest I\'ve seen in their ability to provide the necessary statistics to manage a call center. If you have a version ready for testing, please let me know (on or off list) and I\'ll get it installed here. Thank you! Joe -Original Message- From: Ben Merrills Sent: Thursday, 14. Oct 2004 9:10 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] (Another) Queue log analyser I\'ve been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Another) Queue log analyser
Very nice work Ben, thanks. Here are some additional thoughts - One segmentation that might be useful would be to add outbound calling activities as a either a separate column or even view. On agent stats, it would be useful to see login/logout stamps, login time, ready/not ready time (if this can be tracked, not sure). If you would like, I can send you some example reports that are used in a typical call center, contact me directly if you would find that helpful. Cheers, Wayne Ben Merrills wrote: I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Digium Hardware Question
Another n00b question.. Realizing they will be all the same ext. What is the maximum qty of phones one TDM400P FXS module will support Or what would be the max REN alowable on that module Again assuming north american usage etc... Thanks John B. ___ 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] Another small suggestion patch
It's nice to be able to define the list of asterisk modules we want to
load from the /etc/sysconfig/zaptel file rather than directly in
/etc/init.d/zaptel. I'm using nufone and don't require anything but the
ztdummy (is the rtc-based module better, anyone?), so that's what I've
put here.
These are based on zaptel-1.0-RC1.
John
--- zaptel-1.0/zaptel.init.ole 2003-07-14 14:25:44.0 -0500
+++ zaptel-1.0/zaptel.init 2004-08-18 16:43:04.0 -0500
@@ -27,9 +27,10 @@
RETVAL=0
-MODULES=torisa tor2 wct4xxp wct1xxp wcfxo wcfxs wcusb
-
-RMODULES=wcusb wcfxs wcfxo wct1xxp wct4xxp tor2 torisa
if [ ${DEBUG} = yes ]; then
ARGS=debug=1
--- zaptel-1.0/zaptel.sysconfig.ole 2002-06-06 18:20:24.0 -0500
+++ zaptel-1.0/zaptel.sysconfig 2004-08-18 16:45:29. -0500
@@ -1,2 +1,8 @@
TELEPHONY=yes
#DEBUG=yes
+
+# define the modules we want loaded/unloaded automatically
+#MODULES=torisa tor2 wct4xxp wct1xxp wcfxo wcfxs wcusb
+MODULES=ztdummy
+#RMODULES=wcusb wcfxs wcfxo wct1xxp wct4xxp tor2 torisa
+RMODULES=ztdummy
___
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Re: [Asterisk-Users] Another Firefly update - now with SRV support
Adam Hart wrote: I've also added support for SIP via TCP and the ability to change the SIP port It complains every time you click OK in the Options page about Changing SIP port requires restart, even if you never looked at the SIP page (and don't even have any SIP networks configured). ___ 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] Another Firefly update - now with SRV support
Kevin P. Fleming wrote: Adam Hart wrote: I've also added support for SIP via TCP and the ability to change the SIP port It complains every time you click OK in the Options page about Changing SIP port requires restart, even if you never looked at the SIP page (and don't even have any SIP networks configured). That's a 'feature' - fixed, new version up ___ 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] Another Firefly update - now with SRV support
Adam Hart wrote: That's a 'feature' - fixed, new version up I found another 'feature' :) Although I couldn't get it to happen a 2nd time, I had rung 18005558355 (via like2fone.com's sip server) and was listening to the news and looking through the options dialog box, got through all the options and hit ok. Audio for the most part was ok, but the computer and firefly became unusable and clicking the off button to shutdown was the only way I got control back, wasn't even able to get the process list up and kill things that way. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ 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] Another Firefly update - now with SRV support
With all the talk of SRV support in Asterisk, I thought I'd add support in Firefly so enjoy. Thanks to Olle for helping me with it, explaining the wonderful world of SIP and SRV to me. There's also an option to disable it (seems to take quite a few DNS lookups for SRV) - warning Duane may hunt you down if you do disable it though :) I've also added support for SIP via TCP and the ability to change the SIP port Yes, it's still version 1.8. Hopefully another little update shortly away too for sip presence. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe have a nice day, Adam ___ 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] Another software like Asterisk?
Hi! I am looking for a software that can work as h.323 - sip gateway other than asterisk and free. Someone can help me? Thanks. Mireia ___ 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] Another software like Asterisk?
Try Vovida's Vocal, i think it does it. Mireia Munoz de jesus wrote: Hi! I am looking for a software that can work as h.323 - sip gateway other than asterisk and free. Someone can help me? Thanks. Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Newbie Question: Does Asterisk allow for a hot failover solution in case of failure?
Hi all; I think I have the capacity issues figured out. My next question is whether I can use asterisk for a redundant solution so that if any hardware failure occurs on the phone switch, a spare PBX can route the new calls. I have not been able to find this in the docs, and IIRC, it is possible with Bayonne. Best Wishes, Chris Travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard
Re: [Asterisk-Users] another
As far as I understand it, daytime is a context? so you just use like [daytime] s,1,blahblah etc [weekend] s,1,blahblahweekend etc [EMAIL PROTECTED] wrote: Matt I understand that bit but How do I express the sound file for after that time period ?? Here is what I need to do include = daytime|9:00-21:00|mo-fri|*|* include = weekend|10:00-19:00|sat-sun|*|* I think the above is correct ?? Bit how do I specify the after hours config ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of matt Sent: Thursday, 18 December 2003 2:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] another [EMAIL PROTECTED] wrote: Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In the context put: include = daytime|9:00-17:00|mo-fri|*|* which will include the daytime context during these hours ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] another
Thanks Regards Mick [weekend] s,1,blahblahweekend etc [EMAIL PROTECTED] wrote: Matt I understand that bit but How do I express the sound file for after that time period ?? Here is what I need to do include = daytime|9:00-21:00|mo-fri|*|* include = weekend|10:00-19:00|sat-sun|*|* I think the above is correct ?? Bit how do I specify the after hours config ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of matt Sent: Thursday, 18 December 2003 2:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] another [EMAIL PROTECTED] wrote: Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In the context put: include = daytime|9:00-17:00|mo-fri|*|* which will include the daytime context during these hours ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] another
Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] another
[EMAIL PROTECTED] wrote: Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In the context put: include = daytime|9:00-17:00|mo-fri|*|* which will include the daytime context during these hours ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] another
Matt I understand that bit but How do I express the sound file for after that time period ?? Here is what I need to do include = daytime|9:00-21:00|mo-fri|*|* include = weekend|10:00-19:00|sat-sun|*|* I think the above is correct ?? Bit how do I specify the after hours config ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of matt Sent: Thursday, 18 December 2003 2:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] another [EMAIL PROTECTED] wrote: Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In the context put: include = daytime|9:00-17:00|mo-fri|*|* which will include the daytime context during these hours ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another audio file
If anyone is interested, I've trimmed one of Allison's recordings down to the single word 'welcome', for use as a generic first message when a line is answered. I've put it up at: http://jhcloos.com/sounds/asterisk/welcome.gsm and will submit it to bugs.digium.com as well. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another * crash
I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes. When I debug sip I get a noisy feedback from SER, and then asterisk crashes. the only debug information asterisk is leaving is segmentation fault, dumping core. anyone got a clue? i'm not running the latest CVS so maybe i should upgrade first... rgds, /staffan kerker -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Skickat: den 30 november 2003 20:04 Till: [EMAIL PROTECTED] Ämne: Re: [Asterisk-Users] asterisk server crashing From the console, I see where the call comes in and I can see where the party from the outside hangs up. The next thing that is said is as follows: libgcc_s.so.1 must be installed for pthread_cancel to work. Now I've taken a look on my system and I do in fact have the libgcc_s.so.1 on my system. The location is as follows: /lib/libgcc_s.so.1 It is part of the libgcc-3.2.2-5 package that I have installed on my system. I'm not a programmer, just a novice so I'm not quite sure how to run a backtrace or where the core file would be located. Thanks for your help so far. AJ On Sun, 30 Nov 2003, Roy Sigurd Karlsbakk wrote: - What's the console output after the crash when starting asterisk with -gvvvc? - After the crash, run a backtrace of the core file and send the output here ...perhaps this should be on the FAQ? ...and perhaps the FAQ should be linked to from asterisk.org? roy On Sunday, Nov 30, 2003, at 14:14 Europe/Oslo, [EMAIL PROTECTED] wrote: I deleted all the asterisk related directories and their subdirectories from /usr/src/ and did a brand new check out of zaptel, zapata, libpri, asterisk-addons and asterisk. AJ On Sat, 29 Nov 2003, Tilghman Lesher wrote: On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] wrote: Quoting [EMAIL PROTECTED]: In the zaptel zapata and libpri directories I executed a make clean and did a cvs update and then ran make install. In the asterisk directory I did a make clean, a cvs update and a make upgrade. So I guess the answer to your question is yes I did take care of the other things as well. At least as far as I can see and as far as I know. AJ I don't know if your situation is the same as mine but I have been burned in the past by assuming that cvs update will provide all the lastest files. It only updates files that have previously been downloaded, soo, if you do not have a file that is now part of zaptel for instance, you will still not have that file. Do a fresh checkout to make sure you have all of the needed files. By the way, zapata is no longer needed. It has been incorporated into one of the others. Perhaps you mean subdirectories? True, 'cvs update' will not typically create new subdirectories, so you can do a 'cvs update -d' to have the update create new subdirectories, as 'cvs checkout' does, but 'cvs update' should create new files (in existing directories) just fine. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another * crash
put the core file into gdb, backtrace it and then we'll have some useful information: # gdb asterisk corefile and issue bt on gdb console or run asterisk directly into gdb : # gdb --args asterisk -vvvgc play with it and when it seg faults, issue a 'bt' command matteo. Il lun, 2003-12-01 alle 08:20, Kerker Staffan ha scritto: I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes. When I debug sip I get a noisy feedback from SER, and then asterisk crashes. the only debug information asterisk is leaving is segmentation fault, dumping core. anyone got a clue? i'm not running the latest CVS so maybe i should upgrade first... rgds, /staffan kerker -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Skickat: den 30 november 2003 20:04 Till: [EMAIL PROTECTED] Ämne: Re: [Asterisk-Users] asterisk server crashing From the console, I see where the call comes in and I can see where the party from the outside hangs up. The next thing that is said is as follows: libgcc_s.so.1 must be installed for pthread_cancel to work. Now I've taken a look on my system and I do in fact have the libgcc_s.so.1 on my system. The location is as follows: /lib/libgcc_s.so.1 It is part of the libgcc-3.2.2-5 package that I have installed on my system. I'm not a programmer, just a novice so I'm not quite sure how to run a backtrace or where the core file would be located. Thanks for your help so far. AJ On Sun, 30 Nov 2003, Roy Sigurd Karlsbakk wrote: - What's the console output after the crash when starting asterisk with -gvvvc? - After the crash, run a backtrace of the core file and send the output here ...perhaps this should be on the FAQ? ...and perhaps the FAQ should be linked to from asterisk.org? roy On Sunday, Nov 30, 2003, at 14:14 Europe/Oslo, [EMAIL PROTECTED] wrote: I deleted all the asterisk related directories and their subdirectories from /usr/src/ and did a brand new check out of zaptel, zapata, libpri, asterisk-addons and asterisk. AJ On Sat, 29 Nov 2003, Tilghman Lesher wrote: On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] wrote: Quoting [EMAIL PROTECTED]: In the zaptel zapata and libpri directories I executed a make clean and did a cvs update and then ran make install. In the asterisk directory I did a make clean, a cvs update and a make upgrade. So I guess the answer to your question is yes I did take care of the other things as well. At least as far as I can see and as far as I know. AJ I don't know if your situation is the same as mine but I have been burned in the past by assuming that cvs update will provide all the lastest files. It only updates files that have previously been downloaded, soo, if you do not have a file that is now part of zaptel for instance, you will still not have that file. Do a fresh checkout to make sure you have all of the needed files. By the way, zapata is no longer needed. It has been incorporated into one of the others. Perhaps you mean subdirectories? True, 'cvs update' will not typically create new subdirectories, so you can do a 'cvs update -d' to have the update create new subdirectories, as 'cvs checkout' does, but 'cvs update' should create new files (in existing directories) just fine. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another newbie question
Thanks Jose/Tom for responding to my Newbie questions. its much clearer now. anyhow on to the next [unrelated question] here's the use case: i will need one machine that will answer incoming calls - store the caller's number [caller ID] and then prompt the caller to answer a question by using the dialpad [e.g. please enter your zip code] and then store all the information in MySQL [or any persistant storage will do] what i got so far: it looks like digium Wildcard X100p will answer the phone and get the caller's number [caller ID] but my remaining question is: can i have the caller respond to question [by pressing the dial pad] and can i store that information [and the caller ID] somewhere? any suggestions would be greatly appreciated. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another newbie question
Look into AGI, there a re some examples out there, but it's very much doable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brez Sent: Monday, November 03, 2003 11:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Another newbie question Thanks Jose/Tom for responding to my Newbie questions. its much clearer now. anyhow on to the next [unrelated question] here's the use case: i will need one machine that will answer incoming calls - store the caller's number [caller ID] and then prompt the caller to answer a question by using the dialpad [e.g. please enter your zip code] and then store all the information in MySQL [or any persistant storage will do] what i got so far: it looks like digium Wildcard X100p will answer the phone and get the caller's number [caller ID] but my remaining question is: can i have the caller respond to question [by pressing the dial pad] and can i store that information [and the caller ID] somewhere? any suggestions would be greatly appreciated. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Segmentation Fault (Recording sound)
== Parsing '/etc/asterisk/adsi.conf':
Found -- Accepting call from '890003' to '185' on channel
27, span 1 -- Executing Answer("Zap/27-1", "") in new
stack -- Executing Record("Zap/27-1",
"soundexampless:mp3") in new stack -- Playing
'beep'WARNING[360468]: File translate.c, Line 128
(ast_translator_build_path): No translator path from UNKN to
ULAWWARNING[360468]: File file.c, Line 218 (ast_writestream): Unable to
translate to format mp3, source format ALAWWARNING[360468]: File
app_record.c, Line 166 (record_exec): Problem writing frameSegmentation
fault
I guess this is pretty explanatory.
Regards
Alex
Re: [Asterisk-Users] Another Segmentation Fault (Recording sound)
On Tue, 2003-10-28 at 04:43, Alexandru Coseru wrote: == Parsing '/etc/asterisk/adsi.conf': Found -- Accepting call from '890003' to '185' on channel 27, span 1 -- Executing Answer(Zap/27-1, ) in new stack -- Executing Record(Zap/27-1, soundexampless:mp3) in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128 (ast_translator_build_path): No translator path from UNKN to ULAW WARNING[360468]: File file.c, Line 218 (ast_writestream): Unable to translate to format mp3, source format ALAW WARNING[360468]: File app_record.c, Line 166 (record_exec): Problem writing frame Segmentation fault I guess this is pretty explanatory. While segfaulting isn't a good thing, I'm pretty sure mp3 recording is not supported. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Segmentation Fault (Recording sound)
This was triggered by the lack of an mp3 encoder. Without a backtrace there's no way to know it's fixed for sure, but if you cvs update it should at least fail cleanly and if not please place a bug in the bug tracker. Mark On Tue, 28 Oct 2003, Alexandru Coseru wrote: == Parsing '/etc/asterisk/adsi.conf': Found -- Accepting call from '890003' to '185' on channel 27, span 1 -- Executing Answer(Zap/27-1, ) in new stack -- Executing Record(Zap/27-1, soundexampless:mp3) in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128 (ast_translator_build_path): No translator path from UNKN to ULAW WARNING[360468]: File file.c, Line 218 (ast_writestream): Unable to translate to format mp3, source format ALAW WARNING[360468]: File app_record.c, Line 166 (record_exec): Problem writing frame Segmentation fault I guess this is pretty explanatory. Regards Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] another newbie question: forwarding delay?
Hi, Most embarrased newbie evere here again. Possibly another daft question. I have the digium starter kit lite, so I've got the single FXO and FXS lines All is working well with local sip phones able to dial other phones, conferencing, MOH (Thanks Asterisrk-users list!) along with the one analogue handset etc etc. The one niggleing problem I have now is this: My Dialplan is set to ring the Analogue handset when there is an incoming PSTN call. Fine. This works well BUT there is an annoying delay: I've left a normal analogue handset on the PSTN line, so we have the asterisk internal analogue and the external analogue next to each other. Ehen there is an incoming PSTN call, the external analogue phone starts ringing about 3 seconds before the internal one. If I answer the call on the external handset (ie not via *), the internal phone keeps ringing for another three seconds or so. I'm pretty sure that I've not got an intentional delay anywhere. Does anyone recognize this problem? Perhaps it's just related to fiddling (answering!) the esxternal line before it hits *? I won't fill up the list with my silly dialplan, but its here if anyone cares to check if I've stuffed it up! https://www.turbotas.co.uk/wiki/index.php?page=TurboTasExtensionsConf Thanks in advance! Toby. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Newbie Question
On 2003-06-27 at 14:24, Chip Mefford ([EMAIL PROTECTED]) wrote: Is anyone actually using * as a primary phone system in a small/medium sized business with more than a dozen stations and a real receptionist who handles calls? As impressed as I am with asterisk, and as happy as we are with it as the basis for our IVR/conferencing application, I don't think it is ready to replace a real PBX for general office use. And it doesn't have to because they can work together. There are a lot of very reasonably priced systems on the used market. For example, we use an Eon Millennium (née ITT 3100) that we picked up fully loaded for a few thousand dollars, and for VoIP/IVR/ACD/VM we connect to an asterisk server through its PRI interface. But the PBX itself provides the standard features like nice feature phones (available refurbished for one-third the price of a Cisco 7960), busy lamp / DSS consoles, and ARS tables, that are nicer than anything you could cobble together easily with asterisk at this point. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Newbie Question
On Sat, 2003-06-28 at 02:10, Jim Gottlieb wrote: On 2003-06-27 at 14:24, Chip Mefford ([EMAIL PROTECTED]) wrote: Is anyone actually using * as a primary phone system in a small/medium sized business with more than a dozen stations and a real receptionist who handles calls? As impressed as I am with asterisk, and as happy as we are with it as the basis for our IVR/conferencing application, I don't think it is ready to replace a real PBX for general office use. I would have to disagree. The only reason I hadn't answered this message before is Chip wanted to know about setups with a receptionist. Our office has been using asterisk as our pbx for over a year now. Granted we are a small office of only 5 people, but it hasn't failed us yet. And it doesn't have to because they can work together. There are a lot of very reasonably priced systems on the used market. For example, we use an Eon Millennium (ne ITT 3100) that we picked up fully loaded for a few thousand dollars, and for VoIP/IVR/ACD/VM we connect to an asterisk server through its PRI interface. But the PBX itself provides the standard features like nice feature phones (available refurbished for one-third the price of a Cisco 7960), busy lamp / DSS consoles, and ARS tables, that are nicer than anything you could cobble together easily with asterisk at this point. When you remove the need for a receptionist and if your IVR is setup up well enough that a caller doesn't need to be transfered usually after connected to a user, then all those features on a fancier phone aren't used. I consider the company we had split from to me fairly average, and all the extra buttons on their Intertel system only makes it more likely to drop a call. I think if you consider the average company and down to home use, then add in those companies that are willing to simplify the phone system, you will see a large amount of people ready for a asterisk system. You point out how asterisk can make headway into the those systems that need more. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Newbie Question
I'm getting ready to give asterisk another shot here. Didn't have a lotta luck last time, about 7-8 months back. I have been scanning the list all this time though, lurking. A question that comes up from time to time, that I have yet to see answered is; Is anyone actually using * as a primary phone system in a small/medium sized business with more than a dozen stations and a real receptionist who handles calls? If so, could you email me so we could chat some? Thanks kindly for any input Take care chipper ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another PRI based question
Your Verion Rep is way out in left filed. DID is about the only way to go on PRI's. And doing it is about 3 lines in the CO switch. DMS 500 Table DN Route: Numbers to OFRT 223 table OFRT 223 Trunk Group Name table trkgrp Trunk Definition. That is all there is to it. Routing a single number to a PRI channel on the other hand is a nightmare. We sell DID's in upstate NY for about $20 a hundred. Some one else said it. Get a new Verizon Rep. I have about 6 PRI's going to * boxes in my CO right next to a DMS-500, so I know for a fact this works quite well. As a note, while ISDN Q.931 does call for a 10 Digit Called Party Number (PRI DID) it is entirely possible to send 3,4,5 etc digits as the Called Party Number. Stick with the 10 digits. More typing, but well worth it. Brian -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: Monday, June 09, 2003 4:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Another PRI based question DID's on Asterisk are seen as extensions. Mark On Sat, 7 Jun 2003 [EMAIL PROTECTED] wrote: In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another PRI based question
You hit the nail on the head in saying the tariffs in this area are whacked. A block of 20 DID numbers increases the cost of the PRI about $220 a month and that is with the configuration of 12 inbound and 11 outbound. If you want the configuration otherwise or more DID numbers it costs even more. The area I service is Maryland and Delaware both the states are crazy when it comes to telco issues. In Maryland the install costs are more, monthly charges being less and just the opposite in Delaware. I can be on 2 different job sites less than 5 minutes apart, separated by state lines only and the difference in the PRI monthly cost is $200. If this is not crazy I don't know what is. The rep in this case seems reasonably well informed. With that being the case, he was trying to be mindful in giving us a cheaper option that did relatively the same thing. As a matter of fact, he said he suggests this to most of his customers versus DID numbers. Consider the odds of that, a Verizon rep trying to help or save the customer some money! :) AJ On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 21:04, [EMAIL PROTECTED] wrote: Well I would have went with DIDs however it really increases the pricing of their plan plus we then have to split the channels up as incoming and outgoing. It gets pretty complicated. They already deliver 10 digits in from what I understand. I will inquire from them whether or not I can set my outbound callerid. Would that be setting the name or the number or both? Also its kind of too late to switch now because I have a time frame in which to have the project complete and getting PRIs in is not a quick process even from Verizon. So it is sure to be more difficult from one of the Clec's. Thanks I wasn't suggesting a different CLEC, just a different rep to deal with. It seems either the tariffs in your area are whacked, or you may be getting ripped. For our latest PRI install we went with Telcove, formerly Adelphia. A 20 block of DIDs costs $4 for our DIDs. We didn't have to split our lines between incoming and outgoing. This is the point of PRI, all the signalling goes on out of band to negotiate the channels. Even when we had EM lines from MCI, we had our DIDs and no splitting of the functions. These are reasons why you either need them to explain why you are being told this, or ask for a new rep that is more experienced. Or possibly see if you can't schedule a meeting with a switch tech that is used to actually configing the switch. On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:19, [EMAIL PROTECTED] wrote: In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ It may be time to ask for a new person to work with. You want DID numbers. You want the DIDs to be delivered as the full length number, 10 digits. This lets you put all your incoming calls into a simple context where you define extensions that direct the incoming phone number to a specific function or internal extension. While them delivering 10 digits may be overkill for DID, it allows you to get DID numbers from different exchanges without any problems. Also you may want to make sure they let you set your callerid number on outbound calls. It is helpfull for my setup since my office phones present the main number for the office. The last DID we have is what I use for my home phone, and it presents the last DIDs number so no one sees my office line as my callerid anymore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another PRI based question
In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another PRI based question
Well I would have went with DIDs however it really increases the pricing of their plan plus we then have to split the channels up as incoming and outgoing. It gets pretty complicated. They already deliver 10 digits in from what I understand. I will inquire from them whether or not I can set my outbound callerid. Would that be setting the name or the number or both? Also its kind of too late to switch now because I have a time frame in which to have the project complete and getting PRIs in is not a quick process even from Verizon. So it is sure to be more difficult from one of the Clec's. Thanks AJ On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:19, [EMAIL PROTECTED] wrote: In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ It may be time to ask for a new person to work with. You want DID numbers. You want the DIDs to be delivered as the full length number, 10 digits. This lets you put all your incoming calls into a simple context where you define extensions that direct the incoming phone number to a specific function or internal extension. While them delivering 10 digits may be overkill for DID, it allows you to get DID numbers from different exchanges without any problems. Also you may want to make sure they let you set your callerid number on outbound calls. It is helpfull for my setup since my office phones present the main number for the office. The last DID we have is what I use for my home phone, and it presents the last DIDs number so no one sees my office line as my callerid anymore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another PRI based question
On Sat, 2003-06-07 at 21:04, [EMAIL PROTECTED] wrote: Well I would have went with DIDs however it really increases the pricing of their plan plus we then have to split the channels up as incoming and outgoing. It gets pretty complicated. They already deliver 10 digits in from what I understand. I will inquire from them whether or not I can set my outbound callerid. Would that be setting the name or the number or both? Also its kind of too late to switch now because I have a time frame in which to have the project complete and getting PRIs in is not a quick process even from Verizon. So it is sure to be more difficult from one of the Clec's. Thanks I wasn't suggesting a different CLEC, just a different rep to deal with. It seems either the tariffs in your area are whacked, or you may be getting ripped. For our latest PRI install we went with Telcove, formerly Adelphia. A 20 block of DIDs costs $4 for our DIDs. We didn't have to split our lines between incoming and outgoing. This is the point of PRI, all the signalling goes on out of band to negotiate the channels. Even when we had EM lines from MCI, we had our DIDs and no splitting of the functions. These are reasons why you either need them to explain why you are being told this, or ask for a new rep that is more experienced. Or possibly see if you can't schedule a meeting with a switch tech that is used to actually configing the switch. On Sat, 7 Jun 2003, Steven Critchfield wrote: On Sat, 2003-06-07 at 19:19, [EMAIL PROTECTED] wrote: In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be assigned to a specific channel. It would hunt for an available channel. What we would like to be able to do is that even though it doesn't come in on a specific channel, still be able to route it directly to a specific extension. The representative at Verizon said that we should be able to do this by having the PBX recognize the digits that come in on the line and route it to the specific extension accordingly. Is there a way to do this in asterisk? Thanks again. AJ It may be time to ask for a new person to work with. You want DID numbers. You want the DIDs to be delivered as the full length number, 10 digits. This lets you put all your incoming calls into a simple context where you define extensions that direct the incoming phone number to a specific function or internal extension. While them delivering 10 digits may be overkill for DID, it allows you to get DID numbers from different exchanges without any problems. Also you may want to make sure they let you set your callerid number on outbound calls. It is helpfull for my setup since my office phones present the main number for the office. The last DID we have is what I use for my home phone, and it presents the last DIDs number so no one sees my office line as my callerid anymore. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
