RE: [Asterisk-Users] Hardware
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcelo Pacheco Sent: 26 October 2004 23:23 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware I have a system with VIA chipsets with one T400P (3FXS,1FXO) and 2 E100P (for testing with a cross over cable between them) I have an AMD system with Via chipsets, and have problems with an X100P as I've seen discussed elsewhere on this list. What I'm trying to establish, is whether the problem lies in the hardware (ie: the physical card does not get on with the motherboard), or whether its driver related (ie: the wxfxo driver does not get on with the system drivers for the Via chipset). The reason I'm asking, is that I'd like to upgrade to a TDM400 card with 1 x FXO and 1 x FXS modules. Obviously if the problem is hardware related, then the TDM should be fine, but if software related, I really don't want (and can't afford) to buy a TDM400 + modules only to find out it has the same problem! Does anyone have a TDM400 in an AMD system with a Via chipset, preferably the one as listed below, that could offer any guidance. The motherboard is an MSI KM2M Combo-L SKT A KM266 with a Duron 1800. Any and all help gratefully received, Cheers, Karl bash $ /sbin/lspci :00:00.0 Host bridge: VIA Technologies, Inc. VT8375 [KM266/KL266] Host Bridge :00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP] :00:07.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 :00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 80) :00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 80) :00:10.2 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 80) :00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82) :00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge :00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) :00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) :00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) :01:00.0 VGA compatible controller: S3 Inc. VT8375 [ProSavage8 KM266/KL266] This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Almost there--Remote connection
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ferguson, Michael Sent: 19 October 2004 16:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Almost there--Remote connection [snip] The * server is behind a firewall and I have opened ports 1-10100 5060 5004 4569 IIRC, SIP uses 1-2 by default. Have you changed this to 1-10100? Cheers, Karl This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Almost there--Remote connection
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ferguson, Michael Sent: 19 October 2004 16:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Almost there--Remote connection Thanks. I think that's Iptables. No? I have a hardware firewall. First, have a peek in rtp.conf and see what it says its using. For example, my (modified) version looks like: ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; rtpstart=15000 rtpend=17000 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deon Rodden Sent: Tuesday, October 19, 2004 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Almost there--Remote connection My firewall script has something to the effect of: # Allow Existing traffic through -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT # Incoming VOIP Ports -A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 2727:2727 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 4569:4569 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j ACCEPT That's for IAX2 and SIP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson Sent: Tuesday, October 19, 2004 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Almost there--Remote connection -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ferguson, Michael Sent: 19 October 2004 16:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Almost there--Remote connection [snip] The * server is behind a firewall and I have opened ports 1-10100 5060 5004 4569 IIRC, SIP uses 1-2 by default. Have you changed this to 1-10100? Cheers, Karl __ __ This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk __ __ ___ 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 __ __ This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk __ __ This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Almost there--Remote connection
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ferguson, Michael Sent: 19 October 2004 18:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Almost there--Remote connection I just realised that I neglected to mention that the remote GS100 phone is sitting behind a firewall also. Do I need to open any outgoing ports on that firewall? Considering that one cannot hear anything from the GS100 IP phone? Yes, both phones will need to have ports 1-2 open (having seen your rtp.conf) if they are going o register with your * server. Mine says rtpstart=1 rtpend=2 This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] X100P red alert
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: 19 October 2004 22:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] X100P red alert Alex van Es wrote: Hi all, I just got my X100P card installed and asterisk keeps on complaining that it cannot create a zap channel. I read somewhere on the internet that the zap card will not work when a phoneline is not plugged in, cause is draws power from the phoneline. Is this correct? Of course eventually I will connect it to a phoneline, but I would just like to know this for sure.. Alex Alex, It may not be able to create a zap channel because the driver isn't loaded. Make sure that you have the kernel modules installed and everything working, etc. Too much to go over right now. Look in the wiki, etc. But you will need to have it connected to the phoneline to make calls over the phone line, of course. The X100p does not get power over the phone line. The PCI bus takes care of that. Whilst the card does get power from the line, it does detect if the line is present. If you have the zaptel module loaded, and the ztcfg runs ok, you should have: bash $ cat /proc/zaptel/1 Span 1: WCFXO/0 Wildcard X101P Board 1 1 WCFXO/0/0 FXSKS (In use) This will say RED ALARM if the card is unable to use the line. Check the cord you have plugged in is the correct cord, and has the correct pins connected. I had this originally, and on swapping the cord all worked fine :) Also: bash $ ls -al /dev/zap/* lr-xr-xr-x 1 root root7 Jan 1 1970 /dev/zap/1 - span1/1 crw-rw-rw- 1 root root 196, 254 Jan 1 1970 /dev/zap/channel crw-rw-rw- 1 root root 196, 0 Jan 1 1970 /dev/zap/ctl crw-rw-rw- 1 root root 196, 255 Jan 1 1970 /dev/zap/pseudo crw-rw-rw- 1 root root 196, 253 Jan 1 1970 /dev/zap/timer /dev/zap/span1: total 0 drwxr-xr-x 1 root root 0 Jan 1 1970 . drwxr-xr-x 1 root root 0 Jan 1 1970 .. crw-rw-rw- 1 root root 196, 1 Jan 1 1970 1 bash $ Hope this helps, Cheers, Karl This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Intel Modem vs Digium Cards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: 12 October 2004 04:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards Wow, that's a really sucky attitude. I would expect *Digium* to tell him to go away and solve his own problems. However, if the user community does that, then this is one of the suckiest user communities I've run across in the free software world, and I've been doing free software for many years. Well, IMHO, I would expect it perfectly reasonable for one of three responses from the 'user' community (in order of likelihood): 1) A resounding non-response 2) A response of Well, get a X101P or TDMx0P and try it 3) Some advice on how to resolve the problem. I think it's a good thing the rest of the user community (IMHO) doesn't appear to follow your lead then! I bought a £20 generic X101P from a local voip hardware supplier, and am rather pleased I did. It works most of the time, but drops calls occasionally. I originally thought this might be due to it being a cheap card, but I now find out that the Digium X100P/X101P is the same card with a heatsink, and is 5 times more expensive. Nice. Additionally, the problem appears to be that the AMD Duron/VIA chipset combination in my server doesn't get along with the X10(0|1)P cards, and so if I'd bought the pricey card, I'd still have problems, only solvable by another purchase, and also be significantly out of pocket. FYI: I found all this out by use of Google, and the user community. I go searching for my solutions, and have not called Digium once, and would not expect them to answer calls on my card. To resolve my problem, I may buy a Sipura-3000, I may buy a Digium TDM card. One I woiuld expect support from Digium for, one I would not. Both I would expect to be able to *ask* a question here, and I would *hope* that people that had heard of what I'm using would respond. I would also hope that people that had ever heard of it would keep quiet rather than waste my bandwidth baiting me as to why I didn't buy their pet hardware choice of the moment. Cheers for now, Karl This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Intel Modem vs Digium Cards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: 12 October 2004 09:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards On Tue, 12 Oct 2004 09:05:02 +0100, Karl Dyson [EMAIL PROTECTED] wrote: It works most of the time, but drops calls occasionally. and: To resolve my problem, I may buy a Sipura-3000 I don't know how much effort you have already made but keep in mind that there are parameters in the Zaptel drivers which can be tweaked to at least reduce false hangups significantly. It doesn't look like false hangups (IMHO). Calls drop mid call, regardles of volume. You can go for a week with no drops, and then get 5 in a day (with fairly consistent usage patterns). Don't worry about effort involved -- it's installed in my house, and I tinker when ever possible, and enjoy it. My wife isn't especially over the moon with the dropped calls, but is a patient woman :) There is also a patch now which allows you to disable the disconnect supervision on outgoing Zaptel calls only because that's the scenario where you are most unlikely to need it. I get dropped calls occasionally on both in and outbound calls, and to aid diagnosis have a phone plugged directly into the line as well now (a splitter now has the connection to the X100P and this additional handset). If an inbound call drops, you can pick up this handset and continue the call, this indicating the X100P is dropping the call. I'd quite like to keep the internal card, and hence like the idea of upgrading to a TDM400 This would allow me to continue to use distinctive ringing, especially as I have a use for it(!), but I don't want to spend money on a TDM11B kit only to find that it doesn't get along with the amd/via combination. Is anyone using a TDM400 of any description (preferably FXO *and* FXS just in case) with an AMD/Via combination? I don't think the Sipura does DRING, so would have to loose that functionality if I go that route. On the plus side, I get failover FXO-FXS on the sipura in the event the * server dies, or in the event of a power failure. Swings and roundabouts! Thanks, Karl This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Zaptel 1.0.0. will not compile
Personally I get zaptel, zapata etc from cvs rather than portage. Check your /usr/src/linux symlink points to the correct place... I got all sorts of grief with gentoo when I forgot to put it back after some playing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: 10 October 2004 21:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel 1.0.0. will not compile OK, I got a little further. I don't know why but after re-emerging zapata finally zaptel will build. It is not working however and I still get too many erros during the build. It's now complaining about: CC [M] /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.o /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:163: warning: `fcstab' defined but not used CC [M] /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/tor2.o CC [M] /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/torisa.o /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/torisa.c:1139: warning: `set_tor_base' defined but not used *** Warning: zt_register [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_transmit [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_receive [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_ec_chunk [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_set_dynamic_ioctl [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_unregister [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_alarm_notify [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_rbsbits [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko] has no CRC! *** Warning: zt_transmit [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] has no CRC! *** Warning: zt_receive [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] has no CRC! *** Warning: zt_unregister [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] has no CRC! *** Warning: zt_register [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] has no CRC! *** Warning: zt_dynamic_unregister [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztd-eth.ko] has no CRC! and many many more On Sun, 10 Oct 2004, Remco Barende wrote: Hi list I am trying to install asterisk on a gentoo box running kernel version Linux version 2.6.8-gentoo-r7 (gcc version 3.3.4 20040623 (Gentoo Linux 3.3.4-r1, ssp-3.3.2-2, pie-8.7.6)) #1 Thu Oct 7 20:24:31 CEST 2004 I tried to install from the provided ebuild for Asterisk 1.0.0 but the compile of the zaptel module fails miserably. Compilation seems to start at first but then generates pages and pages of errors like : include/linux/types.h:18: error: syntax error before __kernel_dev_t include/linux/types.h:18: warning: type defaults to `int' in declaration of `__kernel_dev_t' include/linux/types.h:18: warning: data definition has no type or storage class include/linux/types.h:21: error: syntax error before dev_t include/linux/types.h:152: warning: type defaults to `int' in declaration of `f_tinode' include/linux/types.h:152: warning: data definition has no type or storage class include/linux/types.h:155: error: syntax error before '}' token In file included from /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:42: include/linux/kernel.h:15:27: asm/byteorder.h: No such file or directory include/linux/kernel.h:16:21: asm/bug.h: No such file or directory In file included from /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:42: include/linux/kernel.h:81: error: syntax error before size_t include/linux/kernel.h:82: warning: function declaration isn't a prototype include/linux/kernel.h:82: warning: conflicting types for built-in function `snprintf' include/linux/kernel.h:83: error: syntax error before size_t include/linux/kernel.h:83: warning: function declaration isn't a prototype include/linux/time.h:145:31: division by zero in #if include/linux/time.h:145:31: division by zero in #if include/linux/time.h:145:31: division by zero in #if include/linux/time.h:145:31: division by zero in #if include/linux/fs.h:356: error: storage size of `bd_sem' isn't known include/linux/fs.h:357: error: storage size of `bd_mount_sem' isn't known include/linux/fs.h:431: error: storage size of `i_atime' isn't known include/linux/fs.h:432: error: storage size of `i_mtime' isn't known Ideas anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To
RE: [Asterisk-Users] Asterisk 1.0 released
Can anyone confirm if the UK callerid patches were incorporated into CVS or this release? I am still using an older version with the patches applied, and they are working fine, but I cannot give up this functionality. Thanks in advance, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol Sent: 23 September 2004 15:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk 1.0 released Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol ___ 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 This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Digium X100P vs Dodgy Ebay X100P
I wouldn't mind a look at the eBay one, perhaps at a smaller res though (1024x768 ?) I've got one of them waiting to be fitted, and I'd be interested to see if it's the same one. -- -S Me too. I'm using a clone X101P I bought from Goods2World a while back and get occasional dropped calls, so would be interested in comparing the two images. If you could stick them on a web server they could be downloaded easily enough (personally I'd prefer the full images). Cheers, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] How to get the Called id with AGI
Hi, I lookup incoming calls in a MySQL DB to try and match up names and display them with the number on the 7905s. Inbound pstn calls land based on dring settings, so I pass (in this case 918) to be added to the display name to indicate which of my inbound numbers was dialled. So, on the 7905 I get 918:Callers Name 01234567890 Or, 918:Unknown 01234567890 if I don't recognise the number. Hope this helps, Karl ### extensions.conf ### [inbound-pstn-1] exten = s/,1,NoOp(${EXTEN}) exten = s/,2,Goto(inbound-pstn-nocli,s,1) exten = s,1,NoOp(DRING:918) exten = s,2,AGI(clilookup.pl,918) exten = s,3,Goto(inbound-pstn,s,3) exten = t,1,Hangup exten = h,1,Hangup ### clilookup.pl ### #!/usr/bin/perl # my $debug = 1; use Asterisk::AGI; use DBI; $AGI = new Asterisk::AGI; my $dsn = 'dbi:mysql:database=asterisk;host=snip'; my $dbuser = 'snip'; my $dbpass = 'snip'; my $dbh = DBI-connect($dsn, $dbuser, $dbpass,{ PrintError = 0, RaiseError = 0, AutoCommit = 0 }); my %input = $AGI-ReadParse(); foreach $thing (sort keys %input) { $res = $input{$thing}; $AGI-verbose(input{$thing} = $res,1) if $debug; } my $id = shift; $id .= : if $id; $AGI-verbose(id = $id,1) if $debug; $cli = $input{callerid}; my $qcli = $dbh-quote($cli); my $sth = $dbh-prepare(qq{select name from numbers where number=$qcli}) || die; $sth-execute || die; my $cliname; if($sth-rows 0) { $cliname = $sth-fetchrow_array; } else { $cliname = Unknown; } $sth-finish; $AGI-set_callerid(\${id}${cliname}\$cli); $dbh-disconnect; exit 0; -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of usedcanon Sent: 10 June 2004 23:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to get the Called id with AGI Hi Angel, I assume you mean CALLERIDNUM (the number part of caller ID), The easiest thing in that case is to pass it as a parameter to your AGI script extex = 500,1,AGI(myscript.py|${CALLERIDNUM}) in your script you just used the argument passed as usual ( I am not a perl expert, so not sure on the syntax there ) Hope this helps. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Behalf Of Angel Diaz Sent: 10 June 2004 22:41 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to get the Called id with AGI Hi all, Is there a way to get the called id (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI-ReadParse(); $callerid = $input{'callerid'}; $AGI-say_digits($callerid); } Thanks in advance, Angel. Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger http://messenger.yahoo.com/ This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] How to get the Called id with AGI
snip Just realised I answered completely the wrong question. Misread called id as caller id. D'Oh. Sorry, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] re: Voicemail and Cisco Phones
On my 7905s I can configure a voicemail number, which in turn activates a Messages softkey. When pressed, it goes to that exten, which in turn is configured to go to VoiceMailMain. Great, works like a charm. However, the phones also have a go to voicemail timeout after which, the phone diverts the call to voicemail via a temporarily moved. This doesn't work, as it diverts the call to the VoiceMailMain exten, and so instead of taking a message, prompts for a password. So I currently have the timeout on the phone set higher than the timeout on the * box, and so * bounces the call to voicemail (correctly) after the required timeout. I'd like to get the feature working via the phones though, as the To VM softkey and DND features do not work corectly as they function in the same way (it seems). Does anyone know if this is something I can fix? Cheers, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Sent: 07 June 2004 13:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] re: Voicemail and Cisco Phones The Cisco 7960 has a softkey called DND which when pressed as the phone is ringing will sack the call to voicemail. If you where using Cisco CME or CM you can forward all calls to Vmail via CLI or GUI. Kurt __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Re: Caller ID with BT CD50
Ok, I can report that I have just (within the last 2 hours, anyway) downloaded the current cvs head for zaptel and asterisk, and applied Tony's current patches downloaded freshly this evening from nodomain. All applied and compiled, and with a tweak to my dring statements after running asterisk in debug, cli and dring are both working :) Calls with no CLI on either number go straight to voicemail without ringing a phone (usually cold callers in my experience), and I lookup the caller id in a mysql db using an AGI written in perl, that also allows me to display which number the caller dialled as well as a friendly name against the cli in the 7905 displays :) Thanks All, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Karl Dyson Sent: 30 May 2004 00:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Caller ID with BT CD50 Certainly Tony's original patch for CID works with my generic X101P (reports itself as an Intel 537 IIRC). I will get around to downloading his new single patch that includes distinctive ringing and testing it in the next couple of days. Cheers for now, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: 29 May 2004 19:06 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Caller ID with BT CD50 In article [EMAIL PROTECTED], Tony Hoyle [EMAIL PROTECTED] wrote: Kevin Walsh wrote: I downloaded the latest version of your patch, from your website, and it works perfectly. I had waited until I had some time available because I thought I'd have to play around with it for a while. Great. Just need to make sure that it still works for US lines and it's all set. There's some debate whether to use this patch or to wait for one that uses line reversal/guard tone detection... there is the slight problem that the X100P can't detect line reversal so it'd mean everyone upgrading their hardware... still, I have what 'works for me' and will continue hosting it for a while whatever happens. Is that the X100P generically, including the X101P? I would include both algorithms - the line reversal one for the hardware that can do it, and your current one for those that can't. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
Just tried to apply the patch: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Cheers, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 28 May 2004 09:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Robert Boardman [EMAIL PROTECTED] wrote: First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? Yes - it does break the distinctive ring detection, but that's easily sorted out. The correct way would be to move the if (p-use_callerid == 2) code within the existing if (p-use_callerid) block, with a couple more if conditionals here and there. The quick way, however, is to apply the attached chan_zap.c hack over the top of Tony Hoyle's great work. In the standard chan_zap.c, you can't have distinctive ring detection unless you also need Caller*ID detection. My hack makes two changes: 1. Changes an else if into an if to get the world = USA Caller*ID code to run. This will waste a little time, but no more than we were wasting anyway, before Tony's patch was applied. 2. Comment out a line of code to ensure that we always answer after the first ring. We need the first ring to give the the distinctive ring code something to work with, of course. It works for me. Hopefully it'll work for you too. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
Oddly, it looks like the changes were made(!?) It might be, having read Tony's reply, that it's because I applied the uk cli patches from Tony and yourself to the stable rather than head branches? I'll try compiling and let you know. Cheers for now, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 28 May 2004 12:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Karl Dyson [EMAIL PROTECTED] wrote: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Could you try applying the changes by hand. There are only two lines to change and it looks as if the first one went through. I'll check my patch to see if I messed up the original or something silly. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
Well compiles and runs OK, but it doesn't identify the dring. I only started playing with it this morning (only realised it *did* dring when I saw your it's broken dring post) This is what I have in zapata.conf dring1=95,0,0 dring1context=inbound-pstn-1 dring2=325,95,0 dring2context=inbound-pstn-2 is this correct for the UK? (I suspect not, and yes, I have dring on my bt line). Cheers, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Karl Dyson Sent: 28 May 2004 12:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Oddly, it looks like the changes were made(!?) It might be, having read Tony's reply, that it's because I applied the uk cli patches from Tony and yourself to the stable rather than head branches? I'll try compiling and let you know. Cheers for now, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 28 May 2004 12:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Karl Dyson [EMAIL PROTECTED] wrote: Just checked out asterisk stable and zaptel, patched using Tony's patches (which worked, and compiled previously) Then got this when applying your patch. bash # cat ../chan_zap.c.diff | patch -p0 patching file channels/chan_zap.c Hunk #1 succeeded at 4642 (offset -148 lines). Hunk #2 FAILED at 4681. 1 out of 2 hunks FAILED -- saving rejects to file channels/chan_zap.c.rej bash # Could you try applying the changes by hand. There are only two lines to change and it looks as if the first one went through. I'll check my patch to see if I messed up the original or something silly. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Conference Server
If you have usb hardware installed, you can use the ztdummy driver (part of the zaptel bits), and you don't need usb hardware if you're using a 2.6 kernel IIRC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of pesb Sent: 27 May 2004 16:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Server Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
Can confirm it works with Generic X101P *BIG* Thank you :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: 26 May 2004 16:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Caller ID with BT CD50 Tony Hoyle [EMAIL PROTECTED] wrote: Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? It's to allow the X100P to output the caller ID. I can confirm that it works for me. I applied the patches, compiled and installed zaptel, compiled and installed Asterisk and it just worked. Well done. I'll abandon my attempt now. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Asterisk firewall config
Ah yes. I too would like to see ip_conntrack_sip :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: 24 May 2004 08:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk firewall config If your firewall has some form of sip inspect then you will not need to leave open the rtp ports. Chris - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 11:11 PM Subject: [Asterisk-Users] Asterisk firewall config The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
Well, I have a USR015630B, which, according to the FAQ supports (UK) CLI. It supports the at #cli command, but no matter what I try, it will not pick up the caller id. Lucky I already had it and didn't buy it soley for this purpose! My caller display unit (unfortunately a CD60 -- which I've opened to look for CD50 like boards, chips etc) picks up CLI no problem (as you'd expect), as do my DECT phones. of course, only when they're plugged into the line, and not when they're plugged into the ATA186. I will be persuading someone I know to swap the USR for a Hayes modem later today to see if that will do it. otherwise out will come the soldering iron, and eBay will see me bidding on CD50s! Cheers, Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: 22 May 2004 23:55 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 Karl Dyson wrote: Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? From the details on http://www.ainslie.org.uk/callerid/cli_faq.htm it sounds like it wouldn't be too hard to implement, however: The only manufacturers that have ever supported BT Caller ID are Pace, Hayes (Europe), and 3Com/US Robotics. It then goes on to state all 3 of those manufactures no longer support it. I wonder if the low cost geographic VOIP numbers support it? Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
I have a USR modem too but it's a brick unfortunately... Pity as it was a nice modem... the right model too so it might have worked. I wonder what it would take to get the zaptel drivers to pick up CID (or even a cheap conexxant modem or something like that) - these soft modems are pretty dumb so I'd expect there to be some scope for frobbing them to do new stuff. Tony As I say, I lent a Hayes modem to mother in law a year ago, but have since upgraded her to a 56k internal, but cunningly left the old kit in her loft (think of it as redundant storage of my old computer crap ;)) so will pick it up later and try it. failing that, I've located a possible Pace candidate Of course, although my wife is happy with the Cisco 7905s that have sprung up around the house, she still likes the cordless DECT units we have, and so they're plugged into an ATA186. Problem is, they no longer display caller id due to the ATA186 not poking it out in BT format I guess. If I were to buy some US cordless handsets would they do the caller id display? Or am I pushing my luck now! (I'm afraid, nice as they look, the Cisco 7920 cordless phones are a bit out of my price range!!) Cheers, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
ATA186 firmware 3.0+ supports more formats. Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine. Oooh Now which setting would I need to check?? I have a Philips Onis DECT system, which does CLI quite happily on the BT line, and I just checked the ATA and its running : ata000e841adb36 Version: v3.0.0 atasip (Build 031210A) MAC: 0.14.132.26.219.54 SerialNumber: INM07491B0E ProductId: ATA186I2 Features: 0x HardwareVersion: 0x0006 0x So. I guess I must have a setting incorrect?? Thanks in advance, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
If you had CID to start with, I'd expect it to work - eg. you wouldn't get it from the POTS line but if a VOIP call came in and the ATA186 retransmitted that in US format then a US handset would pick it up. But once I can get the cid working, I'm hoping I can persuade the DECT units to pick it up obviously the 7905s display CID happily. but currently only from each other, or incoming IAX calls. I always wondered why phones tended to blip a second before starting to ring these days.. now I know). Me too. suddenly dawned on me recently that old phones would blip due to the CLI coming through moments before the ring This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administr at ion_guide_chapter09186a00801e0eb0.html#wp1113416 ETSI method (type 2). I know my CD50 still doesn't like this method, but then as the first generation device it can be *very* fussy at the best of times - indeed, I think very short line lengths were a problem with this device, they have problems with ISDN where the local NTE actually generates CDS. But as said the BT DECT 3010 is fine, and so is my CD20 etc. Thanks for that I had only really thought of it while typing one of my earlier replies about inbound CLI, and has automatically assumed that the ata186 would not do CLI in a BT/UK way! It looks like the Philips units don't like it anyway (they are ~4 yrs old, mind). I've tried 19e62 and 19e66 so far (the main bits should be fine as defaults I think, and so I just tried getting it to send it before ringing, and after the 1st ring). I'll try the BT CD 60 unit I have in a while to see if that can see the CLI coming from the ATA. Cheers, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Caller ID with BT CD50
Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely, due to the seeming difficulties finding a modem that will *definitely* do it, or the CD50 mod. Which brings me to my question (finally) Has anyone done this [the CD50] mod? It seems the CD50 can be found for a few quid, and I'm not afraid of my soldering iron... I just wondered how people in the UK were capturing callerid. there is so much more you can get asterisk to do if you have access to callerid info. By the way. much as I'd like to do this by switching to ISDN, and be done with it, this server is at my home for me to play with, and ISDN is *not* cheap in fact it would roughly double my quarterly phone bill. other than the price, ISDN would be my solution of choice! Thanks in advance, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Asterisk firewall config
I personally only allow IAX2 in and out from my asterisk box, due to the simplicity of one (udp) port. I do not relish the thought of trying to open the port ranges for SIP securely! As long as your inbound stuff in iax.conf lands in a sensible context, inbound connections would only be able to call your internal extensions, and not make cost calls. Hope that helps Karl -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tony Hoyle Sent: 22 May 2004 23:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk firewall config The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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