Re: [asterisk-users] Problem
On Fri, Jul 20, 2007 at 07:45:32PM -0500, Walter Willis wrote: look my zapata.conf [channels] context=default switchtype=national signalling=fxs_ks Power denial will be identified as a hangup. rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.0 txgain=1.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=3 answeronpolarityswitch=yes hanguponpolarityswitch=yes Those two are meaningless, as your cards will not send polarity reversal events. polarityonanswerdelay=1 callprogress=no musiconhold=default channel = 1,2 add to line: busydetect=yes busycount=3 but the situation is iqual Where have you added those lines? Before the channel line, I hope. have much that it is a clon x100p ??? # ztcfg -v Zaptel Version: 1.4.4 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. it is in debian. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Has anybody used fanless computers of logic supply with asterisk?
Zeeshan Zakaria wrote: Darrick, can you tell which mini-itx board you have and what processor it has on it? I don't them with Pentium processors, instead they have some VIA C3 and C7 processors, which are completely new to me and I have no idea how will they perform with Asterisk. I have a VIA C3 (PD1) system that I've been using for a few years. The C3 is a processor that started with Cyrix. VIA bought them out. The C7 processor is the successor to the C3. The C3 was discontinued because Intel would not renew a license agreement with VIA. The C7's have pretty much full i686 compatibility while the C3 is missing a few of the optimizations (it's fully compatible with i586). Astlinux has an image built specifically for the VIA boards even with some support for the Padlock feature (hardware crypto engine). These systems have performed quite well. They are low power and compact. Not sure what else you need to know. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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] Problem X100P - clone asterisk 1.4.8 no hangup
install asterisk with x100p clone; the problem is that call me and hangup but the interface zap not detect the hangup and the line open. the error is : The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11121 process_zap: Ignoring switchtype [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11121 process_zap: Ignoring rxwink [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11121 process_zap: Ignoring signalling [Jul 22 02:53:22] ERROR[2600]: chan_zap.c:10472 build_channels: Unable to reconfigure channel '1,2' [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11437 reload: Reload of chan_zap.so is unsuccessful! The config file is : zaptel.conf loadzone = us defaultzone=us fxsks=1,2 zapata.conf [channels] context=default switchtype=national rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=4 signalling = fxs_ks channel = 1,2 any idea??? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...
Alvaro Parres wrote: Search at mfcr2.c this: case MFCR2_PROT_MEXICO: And add the next line after that line: mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12; This will help you on calls that have the restricted flag on the ANI only. (Nextel). But not on no caller id calls. I don't know if steve can help us whit the case where no caller id is send. Is that change appropriate for all Mexican operators, or just some? For the time being I have added that to my latest source code. What happens when there is no ANI, and is it the same for all operators in Mexico? Can you send a log of such a call with loglevel=255 in unicall.conf, and I will try to sort this out. It seems strange I haven't had more feedback on this, since there seem to be quite a few users in Mexico. Regards, Steve ___ --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] Has anybody used fanless computers of logic supply with asterisk?
On Sun, Jul 22, 2007 at 02:49:43AM -0500, Darrick Hartman (lists) wrote: Zeeshan Zakaria wrote: Darrick, can you tell which mini-itx board you have and what processor it has on it? I don't them with Pentium processors, instead they have some VIA C3 and C7 processors, which are completely new to me and I have no idea how will they perform with Asterisk. I have a VIA C3 (PD1) system that I've been using for a few years. The C3 is a processor that started with Cyrix. C3 is Centaur. The whole unit is still the original Centaur unit in VIA, I believe (/proc/cpuinfo shows CentaurHauls and not CyrixInstead) VIA bought them out. The C7 processor is the successor to the C3. The C3 was discontinued because Intel would not renew a license agreement with VIA. The C7's have pretty much full i686 compatibility while the C3 is missing a few of the optimizations (it's fully compatible with i586). There are several types of C7, and they differ by their instruction set as well. The kernel has a C3 and C3-2 target. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Problem X100P - clone asterisk 1.4.8 no hangup
Please keep your posts in the same threads. Also: On Sun, Jul 22, 2007 at 02:58:13AM -0500, Walter Willis wrote: install asterisk with x100p clone; the problem is that call me and hangup but the interface zap not detect the hangup and the line open. the error is : The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11121 process_zap: Ignoring switchtype Quite silly to set a PRI switchtype. It is ignored for analog cards anyway. But do remove that one, it is totally unnecssary for you. [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11121 process_zap: Ignoring rxwink Right. So this change of the rxwink timing is ignored. Big deal. [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11121 process_zap: Ignoring signalling Signalling will be left as it was originally. busydetect and busycount weren't ignored. If you really want to be sure of that, restart asterisk. [Jul 22 02:53:22] ERROR[2600]: chan_zap.c:10472 build_channels: Unable to reconfigure channel '1,2' [Jul 22 02:53:22] WARNING[2600]: chan_zap.c:11437 reload: Reload of chan_zap.so is unsuccessful! The config file is : zaptel.conf loadzone = us defaultzone=us fxsks=1,2 zapata.conf [channels] context=default switchtype=national rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=4 signalling = fxs_ks channel = 1,2 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-1.2.22 DeadAGI Hangup
Hi I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Please help. thanks arun ___ --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] Best and easiest soft phone for my Dad..
WipeOut wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. Thats for all the replies to my question.. I will have to check them all out and see what works best for him.. Can anyone post a sample of whats needed in iax.conf for an IAX UA to be able to make and receive calls? Thanks.. ___ --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] IMAP and ODBC voicemail storage
Hi, I'm wondering whether or not I should go for ODBC or IMAP voicemail storage. Before diving into details, I would be very pleased to get input form others. 1. With IMAP, is it necessary to save a copy of voicemails in /var/log files so that a user can still listen to his (or her) own voicemails with his own hardphone ? 2. How then, can you make sure to skip non-voice mails stored in the same email repository ? Regards ___ --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] IMAP and ODBC voicemail storage
Olivier wrote: 1. With IMAP, is it necessary to save a copy of voicemails in /var/log files so that a user can still listen to his (or her) own voicemails with his own hardphone ? no listening your voicemails are only stored in the IMAP folder and accessed from both the email client and the phones. The cool thing is that they are only marked read/deleted/... once so you listen to a message on the phone and your corresponding email is instantly marked read. 2. How then, can you make sure to skip non-voice mails stored in the same email repository ? I usually put voicemail in a separate imap folder but I am sure it also works with only one inbox. Whether it's a voice mail or a regular email can easily be detected by looking at the message headers. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --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] Music on Hold and Announcements
Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! -- Otis ___ --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] IMAP and ODBC voicemail storage
2007/7/22, Stefan Reuter [EMAIL PROTECTED]: Olivier wrote: 1. With IMAP, is it necessary to save a copy of voicemails in /var/log files so that a user can still listen to his (or her) own voicemails with his own hardphone ? no listening your voicemails are only stored in the IMAP folder and accessed from both the email client and the phones. The cool thing is that they are only marked read/deleted/... once so you listen to a message on the phone and your corresponding email is instantly marked read. 2. How then, can you make sure to skip non-voice mails stored in the same email repository ? I usually put voicemail in a separate imap folder but I am sure it also works with only one inbox. Whether it's a voice mail or a regular email can easily be detected by looking at the message headers. Do you mean it is possible (in voicemail.conf) to specify how to look at the message headers ? Here (http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf), I can see how to customize voicemail sending with mailcmd but how would you teach Asterisk to read messages with specific header ? Another set of questions : 3. Let's say you're browsing your incoming emails with your favorite email client. You've got some voicemails among them but you don't want to disturb your neighbours listing to them with your PC speakers. How would you forward the voicemail audio files to your desktop phone ? Calling your own voicemail is an obvious way to listen to those files but I'm wondering if there is a better way to do it 4. I've never heard of email software (client or server) accessing an ODBC storage. Does it exist or shall I understand that voicemail ODBC storage is mainly here to ease custom web application development ? =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
Marc, I like your MediaX Phone ( IAX softphone ), I have been using IDEFISK (Zoiper), but I found your softphone easier to configure. It is stable and simpler to use. Keep up the good work. -baji. -- On 7/21/07, Time Bandit wrote: So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. You don't specify if he's on Windows, Linux or OSX. But if he is on Windows, you can try my softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php There is a version using INI file, so you can put all the settings then zip it and send it to him already configured. hth ___ ___ --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] te110p stays in red alarm
Folks, I installed a te110p to connect to an E-1. Ensured the jumper is on for E-1, installed the card, and got the following from the phone company: hdb3 encoding (verbally confirmed ccs) euroisdn switchtype pri signalling So I set up zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 unused=17-31 dchan=16 loadzone=us defaultzone=us Card tries to come up but goes into RED alarm. Occasionally cycles YELLOW/RED and or RED/RECOVERING or YELLOW/RED/RECOVERING, but always returns to RED. Insert the wcte11xp module with a debug=9 option and get the following in my kernel log: kernel: Expecting top ca1e, got CW Panama says the switch I'm connecting to is an Erickson AXE 10. Thoughts? Thanx, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --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] Wake-Up Call didn't work
I have setup wake up call in * ( 1.2crc1) following those instructions http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP i can enter the time after dialing 77 , and i see there is wakeup files in /tmp but * nevers make the wakeup call when it is due , what can be the problem ? what shall i check? Mario ___ --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] Best and easiest soft phone for my Dad..
Can anyone post a sample of whats needed in iax.conf for an IAX UA to be able to make and receive calls? [7011] type=friend secret=S0m3S3cur3P4ssw0rd qualify=no notransfer=yes [EMAIL PROTECTED] host=dynamic disallow=all allow=ulaw,alaw,gsm context=from-internal callerid=Marc Charbonneau 7011 hth ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CTI interface to control legacy PBX
Hello, I am looking for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 ___ --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] MAKE Menuselect
Does anyone know a way in Asterisk 1.4 to select the options from the menuselect menu from the command line? ___ --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] MAKE Menuselect
kevin, make menuselect - creates an xml file... let me look to see where it is [EMAIL PROTECTED] asterisk-1.4.5]# ls -l menu* Current Directory is /usr/local/src/asterisk-1.4.5 -rw-r--r-- 1 root 2065 Jun 25 18:36 menuselect.makedeps -rw-r--r-- 1 root 1654 Jun 25 18:36 menuselect.makeopts -rw-r--r-- 1 root 37350 Jun 25 18:34 *menuselect-tree* look in menuselect-tree, and... hmm... this looks promising for trying to figure it out... Current Directory is /usr/local/src/asterisk-1.4.5/menuselect -rw-r--r-- 1 root 31131 Aug 19 2006 example_menuselect-tree daveC Kevin Kiely wrote: Does anyone know a way in Asterisk 1.4 to select the options from the menuselect menu from the command line? ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] Debian etch and web voice mail - how to configure it?
Hi Everyone... I am running Asterisk 1.2.13 on Debian Etch. I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says It Works! but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the docs directory there is just the change log and googling it has not helped. Can someone give me a hint as to how to configure this or else point me at some docs? Thanks very much. ___ --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] Debian etch and web voice mail - how to configure it?
jim, the asterisk gui doesn't interact with apache or apache2... it has it's own httpd... perhaps you can move the vmail.cgi script to the apache2 directory structure cgi-bin. I haven't tried that as of yet so I don't know how that would work. daveC Jim Archer wrote: Hi Everyone... I am running Asterisk 1.2.13 on Debian Etch. I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says It Works! but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the docs directory there is just the change log and googling it has not helped. Can someone give me a hint as to how to configure this or else point me at some docs? Thanks very much. ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] Debian etch and web voice mail - how to configure it?
On Sun, Jul 22, 2007 at 12:45:21PM -0400, Jim Archer wrote: Hi Everyone... I am running Asterisk 1.2.13 on Debian Etch. I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. We're talking about http://packages.debian.org/stable/comm/asterisk-web-vmail ( http://packages.debian.org/asterisk-web-vmail ) It actually requires httpd-cgi. Apache happens to be one of the packages that provide that... When I point my browser at my PBX machine, the web page says It Works! but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the docs directory there is just the change log and googling it has not helped. What web page, exactly? That package doesn't have many files: http://packages.debian.org/cgi-bin/search_contents.pl?searchmode=filelistword=asterisk-web-vmailversion=stablearch=all The only relevant one is: /usr/lib/cgi-bin/asterisk/vmail.cgi So what do you get when you try: http://yourhost/cgi-bin/asteriskvmail.cgi -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] MAKE Menuselect
On Sun, Jul 22, 2007 at 12:23:02PM -0400, dave cantera wrote: kevin, make menuselect - creates an xml file... let me look to see where it is [EMAIL PROTECTED] asterisk-1.4.5]# ls -l menu* Current Directory is /usr/local/src/asterisk-1.4.5 -rw-r--r-- 1 root 2065 Jun 25 18:36 menuselect.makedeps -rw-r--r-- 1 root 1654 Jun 25 18:36 menuselect.makeopts -rw-r--r-- 1 root 37350 Jun 25 18:34 *menuselect-tree* look in menuselect-tree, and... hmm... this looks promising for trying to figure it out... Current Directory is /usr/local/src/asterisk-1.4.5/menuselect -rw-r--r-- 1 root 31131 Aug 19 2006 example_menuselect-tree daveC That tree is actually the input for menuselect. Menuselect and command-line are not the best of friends. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Debian etch and web voice mail - how to configure it?
--On Sunday, July 22, 2007 1:17 PM -0400 dave cantera [EMAIL PROTECTED] wrote: the asterisk gui doesn't interact with apache or apache2... it has it's own httpd... perhaps you can move the vmail.cgi script to the apache2 directory structure cgi-bin. I haven't tried that as of yet so I don't know how that would work. Hi Dave, thanks very much. Well I have no burning desire to use Apache at all. The Debian package for web voice mail installed it. I assumed it was required since the package manager included it. If I don't need it, great. One less thing to maintain. But, how do I activate the http server in asterisk then? ___ --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] DTMF recognition problem with PSTN
Hello everyone, I have problem with DTMF recognition when calling from PSTN, my Asterisk box won't read DTMF tone at all. I've tried use cellphone, normal telephone and voip lines, nothing worked. softphone to softphone within extensions are ok. I'm a newbie at this, can anyone point me out where to look? I'd really appreciated. Thanks a lot Nate___ --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] Debian etch and web voice mail - how to configure it?
--On Sunday, July 22, 2007 8:35 PM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: We're talking about http://packages.debian.org/stable/comm/asterisk-web-vmail ( http://packages.debian.org/asterisk-web-vmail ) It actually requires httpd-cgi. Apache happens to be one of the packages that provide that... Ah, okay. When I point my browser at my PBX machine, the web page says It Works! What web page, exactly? It seems like the sample page bundled with the package, but it's not the typical Apache was just installed page I am used to seeing. It really is nothing more than a header that says It Works! which I thought odd. That package doesn't have many files: Right, but it has many dependencies, like Apache 2. So lots of stuff got dragged in along the way. The only relevant one is: /usr/lib/cgi-bin/asterisk/vmail.cgi That's there, but I was not sure if it came with the asterisk package or the asterisk-web-vmail package. So what do you get when you try: http://yourhost/cgi-bin/asteriskvmail.cgi I got NOT FOUND: The requested URL /cgi-bin/asteriskvmail.cgi was not found on this server ___ --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] Debian etch and web voice mail - how to configure it?
A typo in my message: On Sun, Jul 22, 2007 at 08:35:05PM +0300, Tzafrir Cohen wrote: So what do you get when you try: http://yourhost/cgi-bin/asteriskvmail.cgi Oops: http://yourhost/cgi-bin/asterisk/vmail.cgi -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Debian etch and web voice mail - how to configure it?
jim, asterisk does not provide an httpd itself... asteriskNOW does provide lightspeedhttpd.. as tzafrir said in his last email, you would have to move the vmail.cgi to the apache2 cgi-bin directory, then write an html page to execute it. I would have to look at the application to give further insight. if the link tzafrir provided is correct, I can do that... just let me know. what I tend to do is install asteriskNOW and then overwrite * with the latest version... doing anything else on that box is quite rough though... daveC Jim Archer wrote: --On Sunday, July 22, 2007 1:17 PM -0400 dave cantera [EMAIL PROTECTED] wrote: the asterisk gui doesn't interact with apache or apache2... it has it's own httpd... perhaps you can move the vmail.cgi script to the apache2 directory structure cgi-bin. I haven't tried that as of yet so I don't know how that would work. Hi Dave, thanks very much. Well I have no burning desire to use Apache at all. The Debian package for web voice mail installed it. I assumed it was required since the package manager included it. If I don't need it, great. One less thing to maintain. But, how do I activate the http server in asterisk then? ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] Debian etch and web voice mail - how to configure it?
On Sun, Jul 22, 2007 at 01:59:45PM -0400, Jim Archer wrote: --On Sunday, July 22, 2007 8:35 PM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: We're talking about http://packages.debian.org/stable/comm/asterisk-web-vmail ( http://packages.debian.org/asterisk-web-vmail ) It actually requires httpd-cgi. Apache happens to be one of the packages that provide that... Ah, okay. When I point my browser at my PBX machine, the web page says It Works! What web page, exactly? It seems like the sample page bundled with the package, but it's not the typical Apache was just installed page I am used to seeing. It really is nothing more than a header that says It Works! which I thought odd. That package doesn't have many files: Right, but it has many dependencies, like Apache 2. So lots of stuff got dragged in along the way. Do you prefer lighttpd, thttpd or anything else? Tell that to apt explicitly. e.g: apt-get install asterisk-web-vmail lighthttpd This will use lighthttpd instead of apache. Both provide httpd-cgi . The only relevant one is: /usr/lib/cgi-bin/asterisk/vmail.cgi That's there, but I was not sure if it came with the asterisk package or the asterisk-web-vmail package. So what do you get when you try: http://yourhost/cgi-bin/asteriskvmail.cgi I got NOT FOUND: The requested URL /cgi-bin/asteriskvmail.cgi was not found on this server Sorry, my typo: http://yourhost/cgi-bin/asterisk/vmail.cgi -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CTI interface to control legacy PBX
You could by crating an application that sits between the Avaya CTI and listens to Asterisk manager interface. What exactly are you trying to accomplish on the Avaya? On 7/22/07, David Hajek [EMAIL PROTECTED] wrote: Hello, I am looking for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian etch and web voice mail - how to configure it?
--On Sunday, July 22, 2007 9:03 PM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: Oops: http://yourhost/cgi-bin/asterisk/vmail.cgi Thanks Tzafrir! That got the script to work. When I try to log in though, I get an odd error: Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 152. Line 152 seems straightforward, so I double checked to make sure the file is present. It is. I flagged it world readable just as a test. Actually, Apache runs as user nobody on Debian, so that may be needed anyhow. Regardless, it didn't fix the problem. Any ideas? Thanks again! ___ --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] Best and easiest soft phone for my Dad..
Time Bandit wrote: Can anyone post a sample of whats needed in iax.conf for an IAX UA to be able to make and receive calls? [7011] type=friend secret=S0m3S3cur3P4ssw0rd qualify=no notransfer=yes [EMAIL PROTECTED] host=dynamic disallow=all allow=ulaw,alaw,gsm context=from-internal callerid=Marc Charbonneau 7011 hth Thanks.. ___ --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] Has anybody used fanless computers of logic supply with asterisk?
I'll be installing FreePBX on top of Asterisk, and also a 4 port FXO card from sangoma. Will this not overload C7? ___ --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] DTMF recognition problem with PSTN
At 12:58 7/22/2007, Nate wrote: Hello everyone, I have problem with DTMF recognition when calling from PSTN, my Asterisk box won't read DTMF tone at all. I've tried use cellphone, normal telephone and voip lines, nothing worked. softphone to softphone within extensions are ok. I'm a newbie at this, can anyone point me out where to look? I'd really appreciated. Thanks a lot Nate SIP-INFO setting works most of the time. ___ --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] MAKE Menuselect
There have been a lot of updates to the asterisk source recently. I thought the only way to additional options from the menuselect was to run the make menuselect and select the 'optional' install items. Is there an easier way to upgrade asterisk without recompiling the new tarball and re-selecting the additional options. -Original Message- From: dave cantera [mailto:[EMAIL PROTECTED] Sent: Sunday, July 22, 2007 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MAKE Menuselect kevin, make menuselect - creates an xml file... let me look to see where it is [EMAIL PROTECTED] asterisk-1.4.5]# ls -l menu* Current Directory is /usr/local/src/asterisk-1.4.5 -rw-r--r-- 1 root 2065 Jun 25 18:36 menuselect.makedeps -rw-r--r-- 1 root 1654 Jun 25 18:36 menuselect.makeopts -rw-r--r-- 1 root 37350 Jun 25 18:34 *menuselect-tree* look in menuselect-tree, and... hmm... this looks promising for trying to figure it out... Current Directory is /usr/local/src/asterisk-1.4.5/menuselect -rw-r--r-- 1 root 31131 Aug 19 2006 example_menuselect-tree daveC Kevin Kiely wrote: Does anyone know a way in Asterisk 1.4 to select the options from the menuselect menu from the command line? ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.12/910 - Release Date: 7/21/2007 3:52 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CTI interface to control legacy PBX
I need Asterisk to tell Avaya which calls we need to record. Avaya is using their NICE call recording suite. Thanks - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 C F wrote: You could by crating an application that sits between the Avaya CTI and listens to Asterisk manager interface. What exactly are you trying to accomplish on the Avaya? On 7/22/07, David Hajek [EMAIL PROTECTED] wrote: Hello, I am looking for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP and ODBC voicemail storage
Olivier wrote: Do you mean it is possible (in voicemail.conf) to specify how to look at the message headers ? Here ( http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf), I can see how to customize voicemail sending with mailcmd but how would you teach Asterisk to read messages with specific header ? You don't need to. Asterisk adds special headers to the email when it puts it into IMAP. Then, when it goes through the folder looking for voicemails, it reads these headers to get information like the callerid of the person who left the voicemail. If these headers aren't present, it knows it is not a valid voicemail message. Another set of questions : 3. Let's say you're browsing your incoming emails with your favorite email client. You've got some voicemails among them but you don't want to disturb your neighbours listing to them with your PC speakers. How would you forward the voicemail audio files to your desktop phone ? Calling your own voicemail is an obvious way to listen to those files but I'm wondering if there is a better way to do it It is by far the easiest way. You may have to be more specific as to what behavior you are looking for. I presume it could be done with some sort of click-to-dial development and special links in the voicemail email messages. 4. I've never heard of email software (client or server) accessing an ODBC storage. Does it exist or shall I understand that voicemail ODBC storage is mainly here to ease custom web application development ? No such email client exists that I know of. Ease of application development is one benefit of ODBC storage, but not the only one. It also gives you a method of sharing the same database of voicemails between different servers. Admittedly, this is not the most efficient way to do this, but it is one way. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] Music on Hold and Announcements
OCOSA ListAcct wrote: Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! We just used Audacity and blended announcements into the mp3 file... -- Warm Regards, Lee ___ --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] Has anybody used fanless computers of logic supply with asterisk?
Take a look at the Intel D201GLY it beats the pants out of any of the C3/C7 systems and uses DDR2 RAM which is dirt cheap. Actually logic supply sells this board, but if you have an account with DH you can get them a little cheaper. On 7/21/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Darrick, can you tell which mini-itx board you have and what processor it has on it? I don't them with Pentium processors, instead they have some VIA C3 and C7 processors, which are completely new to me and I have no idea how will they perform with Asterisk. On 7/21/07, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? The fanless mini-itx boards should be just fine. There are too many factors to give you a definite answer, but I currently use one with a TDM400 card. A majority of the calls on the board are sip with no transcoding so there is a very small load on the system (hardly noticeable). If you are doing a ton of transcoding or recording calls, your results may be different. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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 -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Viable Alternatives to TDM400P
I have now within 18 months had a second TDM400P die, the first time was random call drops, and now it will not go off hook when making a call. To summarise, the card stopped making calls, I replaced the computer hardware, installed new OS and new Asterisk (from 1.2 to 1.4) without making a difference, the only factor in common is the TDM400P ... oh the card will receive calls just fine, so it's not a surge that has blown anything. Anyway, are there any viable alternatives to the Digium cards for analogue termination as yet. I need a minimum of 3 FXO ports. Are the Sangoma cards any good (I noticed a 5 year warranty on those ones)? Regards, Michael. ___ --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] Has anybody used fanless computers of logicsupply with asterisk?
Andrew, Why do you think the D201GLY at 533mhz are a better board? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Sunday, 22 July 2007 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Has anybody used fanless computers of logicsupply with asterisk? Take a look at the Intel D201GLY it beats the pants out of any of the C3/C7 systems and uses DDR2 RAM which is dirt cheap. Actually logic supply sells this board, but if you have an account with DH you can get them a little cheaper. On 7/21/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Darrick, can you tell which mini-itx board you have and what processor it has on it? I don't them with Pentium processors, instead they have some VIA C3 and C7 processors, which are completely new to me and I have no idea how will they perform with Asterisk. On 7/21/07, Darrick Hartman (lists) [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? The fanless mini-itx boards should be just fine. There are too many factors to give you a definite answer, but I currently use one with a TDM400 card. A majority of the calls on the board are sip with no transcoding so there is a very small load on the system (hardly noticeable). If you are doing a ton of transcoding or recording calls, your results may be different. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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 -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any plans for proper faxing support
Hi Guys, Sounds great, one thing I have noticed with the T.38 passthrough is that it only seems to support 9600. Has anybody else seen this/found a workaround to enable full 14,400. Cheers Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Friday, 20 July 2007 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Any plans for proper faxing support Andrew Joakimsen wrote: I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. You can use Asterisk 1.4's T.38 pass-through support in combination with the new OPAL-using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Lee. ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.12/910 - Release Date: 21/07/2007 3:52 PM -- I am using the free version of SPAMfighter for private users. It has removed 20710 spam emails to date. Paying users do not have this message in their emails. Get the free SPAMfighter here: http://www.spamfighter.com/len No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.14/912 - Release Date: 22/07/2007 7:02 PM ___ --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] Music on Hold and Announcements
OCOSA ListAcct wrote: Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! Interesting question. I actually have some code that will almost do this sitting in a branch. The code actually started out as a joke, but I think I could make it more generic to where it could be useful. Right now, I have two modules - res_monkeys and app_monkeys. If you load res_monkeys on a system, it will pick a random active channel on the system once per minute and play the tt-weasels file to them. This would be a nice module to load on April 1st. :) app_monkeys gives you a dialplan application called Monkeys(). Once you run this on a channel, it will hear the tt-weasels file once a minute for the rest of its lifetime in the system, while executing other applications. I could probably make app_monkeys more generic so that you can specify a frequency and which sound file to play. The one thing you can't do with it is turn this periodic announcement back off. I think I could add it, though ... Anyway, this would only be for 1.6 unless enough people think its useful. Then, I might maintain an unofficial backport to 1.4. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] CAS signalling and FAX solution
I am trying to solve the fax problem by installing an E1 channelbank (Megaplex MP-104) It's a box that has 8 x FXS ports and a single E1 port. The plan was to use one of my 4 E1 ports to connect to the Telstra onramp and one to the MP104. I have since discovered however that the MP104 only supports CAS signalling and I am having trouble getting asterisk to work in this mode. Currently we have Onramp (Australian PRI) port 1 of T410P --- asterisk --- port 2 or T410P --- MP104 In zaptel.conf I have span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,cas,hdb3 cas=32-46:1101 cas=48-62:1101 And this seems to not complain. Some examples ive found on the net have a dchan=47 but I then get a HDLC error from ztcfg In zapata.conf I have group 0 (its working so I won't list it here) And for group 1 group=1 context=from-group1 pridialplan=local signalling=em_e1 overlapdial=yes callerid=asreceived ;channel = 32-39 cas:32-46, 48-62 When I have the channel options in there, it fails to load chan_zap.so with an error. The signalling I would have assumed should be pri_net as I want to supply clock etc but on trying it, the outcome is the same. Either way, I cannot get the MP104 to connect. I CAN however get it to stop reporting errors on its console but I cann't seem to address it from asterisk. When I dial(zap/32/1234) it just goes out g0 and not over the channel ive requested. If anyone has done anything similar, I would really appreciate sample configs or pointing in the right direction. Thanks. -- Kevin Withnall http://kevin.withnall.com/ ILB Computing http://www.ilb.com.au PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 Please consider the environment before printing this e-mail ___ --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] Has anybody used fanless computers of logic supply with asterisk?
Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. For what it's worth, you are not forced to use the GUI that is distributed on Digium's Asterisk appliance. You can enable ssh on the unit and configure it as your normally would. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] Music on Hold and Announcements
Wow seems a bit much? I use 1.2.22.yeah if you make it generic it would be nice and I would probably upgrade. I guess. The only other way to do this is to just drop the announcements and record a message on hold for a specific group with music in the background at the recording time. So for team 1 context [team1] and team 2 context [team2] and play various messages specific for the groups. Russell your a genius.nice setup. Otis Russell Bryant wrote: OCOSA ListAcct wrote: Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! Interesting question. I actually have some code that will almost do this sitting in a branch. The code actually started out as a joke, but I think I could make it more generic to where it could be useful. Right now, I have two modules - res_monkeys and app_monkeys. If you load res_monkeys on a system, it will pick a random active channel on the system once per minute and play the tt-weasels file to them. This would be a nice module to load on April 1st. :) app_monkeys gives you a dialplan application called Monkeys(). Once you run this on a channel, it will hear the tt-weasels file once a minute for the rest of its lifetime in the system, while executing other applications. I could probably make app_monkeys more generic so that you can specify a frequency and which sound file to play. The one thing you can't do with it is turn this periodic announcement back off. I think I could add it, though ... Anyway, this would only be for 1.6 unless enough people think its useful. Then, I might maintain an unofficial backport to 1.4. ___ --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] IP Trunk between Asterisk and another IP PBX
Dears; If I need to do an SIP Trunk between Asterisk and another IP PBX, then no need to do registeration to that IP PBX (it the other IP PBX support this)? In this case, do I need to make the host an static IP address? Or what is the method to determine that no registeration? From the other side, that is the relation between making the qualify = yes and the other IP PBX? Why we make qualify = yes if other IP PBX does ot support registeration for SIP? Regards, --- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --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] Upgrade and keep the configuration
Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. Regards, -- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --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] extension.conf doesn't reload?
Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Has anybody used fanless computers of logic supply with asterisk?
On Sun, Jul 22, 2007 at 04:45:05PM -0400, Zeeshan Zakaria wrote: I'll be installing FreePBX on top of Asterisk, and also a 4 port FXO card from sangoma. Will this not overload C7? You didn't mention the exact processor (cat /proc/cpuinfo ). But generally 4 concurrent uncompressed calls are well below its limits. The overhead for Zptel is not that large. We regularily put many more analog channels on a similar machine. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Debian etch and web voice mail - how to configure it?
On Sun, Jul 22, 2007 at 02:02:54PM -0400, dave cantera wrote: jim, asterisk does not provide an httpd itself... asteriskNOW does provide lightspeedhttpd.. as tzafrir said in his last email, you would have to move the vmail.cgi to the apache2 cgi-bin directory, I did *not* say such a thing (or even write one). Debian has offers more than one or two HTTPDs. Hence it has to provide a policy also on where packages place content to be served by those HTTPDs. http://www.us.debian.org/doc/debian-policy/ch-customized-programs.html#s-web-appl I have just made a small typo there. then write an html page to execute it. I would have to look at the application to give further insight. if the link tzafrir provided is correct, I can do that... just let me know. what I tend to do is install asteriskNOW and then overwrite * with the latest version... doing anything else on that box is quite rough though... daveC AstriskNow is not something you can install on top of you existing Extch installation, as it is a complete distribution. You probably refer to the asterisk-gui. There is a work-in-progress package on pkg-voip for asterisk-gui through I have other things to do. If anybody wants to pick it up, be my guest... However, try the package ari. As unmaintained as vmail.cgi, but at least looks nicer. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Debian etch and web voice mail - how to configure it?
On Sun, Jul 22, 2007 at 03:06:35PM -0400, Jim Archer wrote: --On Sunday, July 22, 2007 9:03 PM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: Oops: http://yourhost/cgi-bin/asterisk/vmail.cgi Thanks Tzafrir! That got the script to work. When I try to log in though, I get an odd error: Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 152. It cannot read that file, or it cannot read /etc/asterisk . I can't think of a solution for this I really like. Basically add the web server to the group asterisk or any other permissions games. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] extension.conf doesn't reload?
Devraj Mukherjee wrote: Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found Try dialplan reload ___ --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] extension.conf doesn't reload?
On 7/23/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found -- Hi when you issue reload command whole asterisk configs are reloaded ( in 1.2.X) but when you reload it says Please use 'module reload' instead may be you try to reload required module ( not tried in 1.4.x) To cross check issue command show dialplan and check your modified config effected or not ram ___ --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] Upgrade and keep the configuration
If you are moving from 1.2.x to 1.4.x then you may need to update a bit of your dial plan. If not you just needs to install the new version of asterisk and remove the modules from the old version and you should be good to go. Also I personally back up all my config filed just in case. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, July 23, 2007 7:15 AM Subject: [asterisk-users] Upgrade and keep the configuration Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. Regards, -- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wake-Up Call didn't work
Can it be that asterisk does not have permission to copy the file over ? Also check your date settings on the server. - Original Message - From: Asterisk guy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, July 22, 2007 5:29 PM Subject: [asterisk-users] Wake-Up Call didn't work I have setup wake up call in * ( 1.2crc1) following those instructions http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP i can enter the time after dialing 77 , and i see there is wakeup files in /tmp but * nevers make the wakeup call when it is due , what can be the problem ? what shall i check? Mario -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade and keep the configuration
On Sun, Jul 22, 2007 at 09:15:30PM -0700, bilal ghayyad wrote: Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. 'make install' of either of those will not override your configuration. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] extension.conf doesn't reload?
On Mon, Jul 23, 2007 at 02:16:57PM +1000, Devraj Mukherjee wrote: pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. It did work. Howver you were warned that the command is deprecated. This means it will be removed in Asterisk 1.6 . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] extension.conf doesn't reload?
Hey Bruce, Thanks for your prompt response. Your suggestion lead to me finding out that the dialplan module was not loaded. I investigated this further and found out that /etc/asterisk/asterisk.conf was looking in /usr/lib/asterisk for modules. My machine is running 64bit CentOS and has all the modules in /usr/lib64/asterisk I modified /etc/asterisk/asterisk.conf to look for modules in that directory and everything works just fine. Thanks again. On 7/23/07, Bruce Ferrell [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found Try dialplan reload ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Music on Hold and Announcements
On Sun, Jul 22, 2007 at 07:10:18PM -0400, Lee Jenkins wrote: OCOSA ListAcct wrote: Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! We just used Audacity and blended announcements into the mp3 file... As a note: you'd probably get better results by converting those to wav (CPU owork needed to play, and probably even disk space used) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Debian etch and web voice mail - how to configure it?
--On Monday, July 23, 2007 7:40 AM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: That got the script to work. When I try to log in though, I get an odd error: Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 152. It cannot read that file, or it cannot read /etc/asterisk . I can't think of a solution for this I really like. Basically add the web server to the group asterisk or any other permissions games. I had to set the /etc/asterisk directory to a+x. There are a variety of other permission issues, which I'll work through. Thanks again for your help. Best, Jim ___ --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
