Re: [asterisk-users] Problem with AGI Script
Hello,
I had a similar problem with a PHP AGI script. I'm not sure if it's a bug
or what, but it seems the new way of setting variables is an application, no
way could I get it to work.
In the end I set a user defined variable in the AGI like this: write(SET
VARIABLE myvariable);
Then in the dial play did something like Set(CALLERID(number)=${myvariable})
It may not be the most elegant solution but it works fine for me.
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: 18 November 2007 16:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with AGI Script
didier wrote:
Callerid(number) ? or callerid(num) ?
Grasshopper, you will find many answers you seek by looking in
/path/to/src/asterisk-1.4/doc/channelvariables.txt
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[asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone -- Asterisk -- Linksys SPA3102 -- PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone -- Asterisk-- Linksys SPA3102 -- PSTN (In Use) -- Use IAX Can any one help me with some dial plan logic for this; I'm confused as to the best way around this? Thanks in advance Chris -- Chris Blunt Entropy IT Ltd ___ --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] Meetme define context
Hi All, I'm still having trouble trying to figure out if it is possible to define (in the dial plan) a context for meetme? I'm using 1.4.4 with dialplan logic of: exten = 123,1,Meetme(,Msa,) This defaults to conferences defined within the rooms context of meetme.conf Is it possible to specify another context as with voicemail? Or can any one think of another way to do this, my ultimate goal is to have only certain conferences available to certain extension numbers. For example, call extension 123 have access to conference numbers ,1112,1113 call extension 124 have access to conferences 1114,1115 etc. Best regards Chris -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme context.
Hi All, Is it possible to specify the context of a meetme conference under 1.4.x? By default all meeting rooms are generated under the context rooms, I would like to use other contexts depending on what extension number is used to call the meetme application. If it is possible can someone post a syntax example for extensions.conf. Best regards Chris -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Zaptel linux26
Hi The various bits of instruction out there on compiling Zaptel on 2.6 seem to be a bit misleading. With the latest versions there is no need to run make linux26 Simply run Configure Make Make install Optionally I believe you can run make menuselect first to choose packages? Hope this helps regards Chris -- Original Message: Date: Tue, 29 May 2007 12:11:55 -0700 From: Khaled Chehab [EMAIL PROTECTED] Subject: [asterisk-users] Zaptel linux26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 I am using centos 4.4 ,when I am compiling zapltel using l make linux26 ,error accrued ,what s missing [EMAIL PROTECTED] zaptel]# make linux26 grep: /include/linux/autoconf.h: No such file or directory make: *** No rule to make target `linux26'. Stop. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
Hi List / Tzafrir I can't thank you enough for your support through this problem. I had another look on voip-info.org/wiki at CentOS. There is a good post on installing Astrerisk on CentOS, I was reading it through, and thought I would double check a few things. It turns out the linux symbolic links to the Kernel source were pointing to the wrong version. Somone else who had been on the server before me had tried to install the source but had not correctly identified it was the smp version required. Using some of the knowledge you had shared with me and doged determination it now works. When people post questions asking what distro to use, pick one and stick to it. I'm certain half of my troubles have arisen from using a distro I am not familier with. Althouh Slackware is considered Hard Core by some, it's what I am more used to (and installing from CD my self). Again, many thanks Chris -- Chris Blunt -Original Message- Date: Tue, 10 Apr 2007 19:56:43 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Apr 04, 2007 at 05:52:46PM +0100, Chris Blunt wrote: Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! maybe ztdummy.ko was not regenerated? 'make clean' is normally not needed when changing kernel versions, as Kbuild is usually smart enough to tell the difference. What is the output of: modinfo ./ztdummy.ko -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Zaptel 1.4.1 Install Modules CentOS
Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! I can't thank you enough for your continued help. Chris -- Chris Blunt -Original Message- yum install kernel-smp-devel I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Yes. No suggested location for the kerenl source. This should be fixed by installing the relevant kernel-devel package (which has a partial copy of the kernel build tree, configured for the specific kernel) -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS
Hi Tzafir / List. Thank you for your reply. I have run: make clean Configure Make Make install I get no compile errors, but still the same problems if I try to insmod zaptel As you suggested I tried modinfo zaptel Which resulted in: modinfo: could not find module zaptel I also tried depmod with the same result and finally I tried insmod ./ztdummy from the src/zaptel-1.4.1 directory which resulted in: insmod: error inserting './ztdummy.ko': -1 Invalid module format Your continued help is much appreciated. Chris Original Message Reads. Message: 8 Date: Tue, 3 Apr 2007 19:57:40 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote: Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. Have you run 'make install'? What is the output of modinfo zaptel Any change if you run: depmod [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory insmod ./ztdummy.ko But it should fail (e.g: because zaptel is not loaded). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com http://www.xorcom.com/ iax:[EMAIL PROTECTED]/tzafrir -- Chris Blunt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: asterisk-users Digest, Vol 33, Issue 15
Hi Tzafir / List Here is some more information obtained from the commands you gave me: 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:21:39 CDT 2006 i686 i686 i386 GNU/Linux kernel-2.6.9-42.EL kernel-smp-2.6.9-42.EL kernel-ib-1.0-1 kernel-devel-2.6.9-42.0.3.EL kernel-2.6.9-42.0.3.EL kernel-smp-2.6.9-42.0.3.EL kernel-utils-2.4-13.1.83 I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Again thanks for your help, I am only a Linux beginner, and even more of a noob with CentOS. Best regards Chris -- Chris Blunt -Original Message- This means that you built the modules vs. a kernel source tree that does not match your running kernel. What kernel do you run? What is the output of uname -a You mentioned you were running on CentOS. Do you have the proper kernel-devel package for your kernel? rpm -qa | grep kernel And while we're at it, let's check the first guess of the makefile for the location of the kernel source tree: ls -l /lib/modules/`uname -r` The build link there should have the information. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! I can't thank you enough for your continued help. Chris -- Chris Blunt -Original Message- yum install kernel-smp-devel I did check the /lib/modules/2.6.9-42.0.3.ELsmp directory but there is no build link, could this be the problem? Yes. No suggested location for the kerenl source. This should be fixed by installing the relevant kernel-devel package (which has a partial copy of the kernel build tree, configured for the specific kernel) -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.1 Install Modules CentOS
Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory This is really causing me to scratch my head, the timer module is loaded ok, I simply don't know what is going wrong with the modules? I'm a bit out of my depth with CentOS, as this isn't my server (I'm a Slackware guy) Any pointers seriously appreciated. Thanks Chris -- Chris Blunt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipura SPA2000 Transfer Call
Hi List, I have a Sipura SPA 2000, and I am trying to get call transfer to work. I am using an old version of Asterisk, and as far as I am aware I have feature.conf disabled in the dialplan (I am happy with this do far). So I am trying to get the SPA to do the transfer. It looks like *98 is the transfer code, but it just seems to ignore this. I read somewhere about having to do a hook flash first, but this is a UK phone, which button would that be? Have I got something in the SPA disabled or just going about it the wrong way? Any pointers appreciated. Chris -- Chris Blunt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with CentOS ztdummy kernel 2.6
Hi List, I am having some trouble with installing the latest version of ztdummy on a CentOS Kernel 2.6 system. I have installed a few Asterisk systems on Slackware Kernel 2.4.x without any issues, unfortunately there is no choice about this distro, or kernel as it has been preinstalled by someone else. And so I am in the dark with an unfamiliar distro and kernel. I am fairly sure the kernel source has been installed. I'm not sure the timer module is installed in the kernel, is it possible to check? If not I think I will need to use ztdummy for definite. Any help with this would be a real life saver. Thanks - Chris From the zaptel-1.2.13 directory I issue the make linux26 command with the following result: make: *** No rule to make target `linux26'. Stop. Just issuing the make command does seem to work and concludes with: make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Make install outputs the following: make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 HOTPLUG_FIRMWARE=yes modules make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Building modules, stage 2. MODPOST make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' build_tools/genudevrules /etc/udev/rules.d/zaptel.rules if [ -d /usr/lib/hotplug/firmware ]; then \ install -m 644 wct4xxp/*.ima /usr/lib/hotplug/firmware; \ install -m 644 wctc4xxp/*.bin /usr/lib/hotplug/firmware; \ fi if [ -d /lib/firmware ]; then \ install -m 644 wct4xxp/*.ima /lib/firmware; \ install -m 644 wctc4xxp/*.bin /lib/firmware; \ fi Installed firmware install -D -m 755 ztcfg /sbin/ztcfg if [ -f sethdlc-new ]; then \ install -D -m 755 sethdlc-new /sbin/sethdlc; \ elif [ -f sethdlc ]; then \ install -D -m 755 sethdlc /sbin/sethdlc ; \ fi if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi for x in zaptel.ko tor2.ko torisa.ko wcusb.ko wcfxo.ko wctdm.ko wctdm24xxp.ko ztdynamic.ko ztd-eth.ko wct1xxp.ko wcte11xp.ko pciradio.ko ztd-loc.ko ztdummy.ko zttranscode.ko; do \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/extra/$x ; \ done; \ make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install; make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' INSTALL /usr/src/zaptel-1.2.13/pciradio.ko INSTALL /usr/src/zaptel-1.2.13/tor2.ko INSTALL /usr/src/zaptel-1.2.13/torisa.ko INSTALL /usr/src/zaptel-1.2.13/wcfxo.ko INSTALL /usr/src/zaptel-1.2.13/wct1xxp.ko INSTALL /usr/src/zaptel-1.2.13/wct4xxp/wct4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctc4xxp/wctc4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctdm.ko INSTALL /usr/src/zaptel-1.2.13/wctdm24xxp.ko INSTALL /usr/src/zaptel-1.2.13/wcte11xp.ko INSTALL /usr/src/zaptel-1.2.13/wcusb.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxo.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxs.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp_usb.ko INSTALL /usr/src/zaptel-1.2.13/zaptel.ko INSTALL /usr/src/zaptel-1.2.13/ztd-eth.ko INSTALL /usr/src/zaptel-1.2.13/ztd-loc.ko INSTALL /usr/src/zaptel-1.2.13/ztdummy.ko INSTALL /usr/src/zaptel-1.2.13/ztdynamic.ko INSTALL /usr/src/zaptel-1.2.13/zttranscode.ko make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' if ! [ -f wcfxsusb.o ]; then \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxsusb.o; \ fi; \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxs.o install -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0 [ `id -u` = 0 ] /sbin/ldconfig || : rm -f /usr/lib/libtonezone.so ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so.1 ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then /sbin/restorecon -v /usr/lib/libtonezone.so; fi install -D -m 644 zaptel.h /usr/include/linux/zaptel.h install -D -m 644 torisa.h /usr/include/linux/torisa.h install -D -m 644 tonezone.h /usr/include/tonezone.h install -m 644 doc/ztcfg.8 /usr/share/man/man8 install -m 644 doc/zttool.8 /usr/share/man/man8 [ `id -u` = 0 ] /sbin/depmod -a 2.6.9-42.0.3.ELsmp || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc wct4xxp wcfxs wctdm8xxp wct2xxp Building /etc/modprobe.conf... Once it is installed I run: modprobe ztdummy with the following result. FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
[asterisk-users] Need quality toll free 800 number over IAX?
Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent autologoff dynamic queue members - Brain aches please help
Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo Ann Boon), call is returned to queue and the cycle continues. If the same agent doesn't answer twice they are logged out and the call is again returned to the queue Now I want the queued call to fall out of the queue if there are no agents logged in. My Googling and searching of the wiki hints at using leavewhenempty=yes Unfortunately this seems to be unsupported when used with agentcallback. Further research suggested using dynamic queue members, where by a queued call addresses the dynamic member directly by channel avoiding the dialplan altogether. I have now tried this approach, but my agents are not being logged off automatically using autologoff=20. Any help to easy my lack of sanity would be greatly appreciated Best regards, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with dial plan - two attempts at calling agent before logging agent off?
Hi List, I'm attempting to set up a queue and agents using agent call back. This is all working fine with the queue and the agents login etc However. In my dial plan I a set variable when a call is entered into the queue to identify the origin of the call, then when the agent is called I test to see if the call is from the queue. If it is, the dial plan does not go to VM if the agent does not answer, it gives BUSY and the call is returned to the queue. The call could well be passed to the same agent again from the queue, which I am okay with - BUT I only want it to try twice before logging the agent out (just in case they have gone AWOL and not logged out). The autologoff=xx in agents.conf doesn't seem to work with agentcallback. I have tried setting another variable as a counter with some logic tests to see the number of attempts to call the agent, but this is failing as the variable appears to be lost when the call goes back to the queue. Can anyone suggest an answer to this puzzle for me. Many thanks Chris -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI PHP Issues (AGI script runs but phone hangs up too quickly)
Sorry to re-post this but I'm sure it's something simple that someone has
found before.
To summarise:
Dial plan answers the phone
AGI script executes
AGI debug in console show phonetics ABC - However no audio at all on the
phone and this step is less than 1 second.
Dial plan Busy
Phone hangs up.
Total time less than a second.
This is such a simple AGI script do I need the PHPAGI Library - this seems
like a sledgehammer to crack a peanut.
Thanks again.
Original post:
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
My script seems to execute ok, and I can see asterisk playing the sounds but
my phone goes from ringing to busy, and I don't hear the phontics.
Below are the relevant bits from my PHP, Console, and extensions.conf.
I would be most grateful if someone could show me the way.
Thanks in advance:
Chris
Asterisk ver: 1.2.10
PHP:
#!/usr/local/php/bin/php -q
?php
$stdin = fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/var/log/asterisk/my_agi.log', 'w');
while (!feof($stdin)) {
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if (($temp == ) || ($temp == \n)) {
break;
}
}
fputs($stdout,SAY PHONETIC \abc\ \#\ \n);
fflush($stdout);
$msg = fgets($stdin,1024);
fputs($stdlog,$msg . \n);
?
Extensions.conf:
exten = 4343,1,Answer
exten = 4343,2,AGI(example.php)
exten = 4343,3,Busy
AGI Debug:
AGI Rx SAY PHONETIC abc #
-- Playing 'phonetic/a_p' (language 'en')
-- Playing 'phonetic/b_p' (language 'en')
-- Playing 'phonetic/c_p' (language 'en')
-- AGI Script example.php completed, returning 0
-- Executing Busy(SIP/4321-081b9498, ) in new stack
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[asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)
Sorry to bother you all with what is probably a simple question.
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
My script seems to execute ok, and I can see asterisk playing the sounds but
my phone goes from ringing to busy, and I don't hear the phontics.
Below are the relevant bit from my PHP, Console, and extensions.conf.
I would be most grateful if someone could show me the way.
Thanks in advance:
Chris
Asterisk ver: 1.2.10
PHP:
#!/usr/local/php/bin/php -q
?php
$stdin = fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/var/log/asterisk/my_agi.log', 'w');
while (!feof($stdin)) {
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if (($temp == ) || ($temp == \n)) {
break;
}
}
fputs($stdout,SAY PHONETIC \abc\ \#\ \n);
fflush($stdout);
$msg = fgets($stdin,1024);
fputs($stdlog,$msg . \n);
?
Extensions.conf:
exten = 4343,1,Answer
exten = 4343,2,AGI(example.php)
exten = 4343,3,Busy
AGI Debug:
AGI Rx SAY PHONETIC abc #
-- Playing 'phonetic/a_p' (language 'en')
-- Playing 'phonetic/b_p' (language 'en')
-- Playing 'phonetic/c_p' (language 'en')
-- AGI Script example.php completed, returning 0
-- Executing Busy(SIP/4321-081b9498, ) in new stack
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[asterisk-users] Is there a smarter way to ban expensive calls in dial plan?
Hi List, I need a bit of advice please. I want to ban calls to expensive destinations such as cell phones. This is fairly simple here in the UK because all cell phone numbers begin with a 7 where as all geographic numbers begin 1 and 2 Elsewhere this is different, take Andorra for example all numbers begin 376, cell phone numbers are 3763, 3764 and 3765 So if I try the following dial plan my pattern always matches the first wild card Exten = _00376.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion I seem to have been able to fix this with adding an x after the 6 in the first extension to make the patterns all the same length and thus making a better match with the blocked numbers. Example: Exten = _00376x.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion This is just so long winded, and you can imagine doing this for a huge list of destinations. If any one can suggest an improved or more efficient way of doing this, I would be greatly appreciated! Best regards Chris -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load balenced (ADSL) network connections, is it possible?
Hi List, I need to put an Asterisk server in a remote office where only ADSL is available. Maximum of 8meg downstream 646k upstream. I need to handle 20 concurrent calls over IAX preferably uLaw, so 64k per channel. Is it possible to somehow have multiple NICs in the server each with a different IP address pointing to a different default gateway (router). But then some how load balanced into a virtual network connection? Any ideas or solutions would be appreciated just in case I have gone off at a wild tangent. Thanks -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan question
Hi List, this is probably quite straightforward I need to call a sip extension for 15 seconds, if unanswered I then need to call the same sip extension and an additional sip extension for a further 15 seconds, finally if the calls arent answered I need it to go to a generic unavailable VM. My question is if the first sip extension is busy, and I dont have the 100 + x busy VM defined will it just carry on to the next priority without complaining or is there a more elegant way of achieving this? Example of my dialplan: exten = 0870xxx,1,Wait(2) exten = 0870xxx,2,Answer() exten = 0870xxx,3,Playback(cust-greeting) exten = 0870xxx,4,SetCIDName(Tech) exten = 0870xxx,5,Dial(SIP/4902,15,tr) exten = 0870xxx,6,Dial(SIP/4902SIP/4903,15,tr) exten = 0870xxx,7,Voicemail(u7003) exten = 0870xxx,8,Hangup Thanks for your time and advice. -- Chris Blunt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrading
Hi List, I was wondering what is the best way to upgrade an Asterisk system to the latest version. I know there is the patch method, but if I am jumping 3 or 4 versions is a re-install the best way? Should I just make the files then manually copy them in? Does this avoid overwriting any modified sound files etc? Should I delete the current files or move / make a copy to a different location first? I know this is a lot of questions but I am hoping for a best practice idea etc Regards Chris -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Configure Voipjet.com content in Asterisk
Hi Chandramouli
Setting up VoipJet is quite simple really, you have done all the hard bit to
get you Asterisk config this far.
Firstly may I point out if you are posting your configuration to this list
you change your password information, as you have just given everyone access
to your account at voipjet.
Make the changes to your iax.conf as voipjet suggest, the config they give
you is generated for you and is not generic.
Then you will need to add some provision in your dialplan (extensions.conf)
to route your outbound calls.
Something like:
exten = _9.,1,SetCIDNum(123456432) ; This is your proper phone number
exten = _9.,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},45,tr) ;dials the number
What this does: To make a call dial 9 followed by the number and press dial
on x-lite. The first command sets your Caller ID number. The second line
strips the 9 from the beginning of your number and hands the call to voipjet
to terminate.
You will need to ensure that your users have access to the context in wich
you put these entries.
As voipjet are US based you will need to dial your numbers in a us format.
Ie. 312 xxx (for calling Chicago).
Hope this helps you out.
Chris
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[Asterisk-Users] Meetme from MySQL
Hi List, Is it possible to store meetme config in a MySQL table? If so, any pointers would be appreciated. Thanks Chris -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for quality inbound DID - IAX providers, UK, USA, Australia
Hi All, I am looking for a provider/s of inbound DID IAX numbers, for UK, USA, and Australia. Preferably free or low cost J Can anyone make a good reference? Many thanks Chris PS: I appreciate this is perhaps a little OT, please feel free to reply off list. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Three way calling with X-Lite / MeetMe
Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesnt look like it supports this. Can this be achieved with MeetMe or AppConference, if it can please tell me how J Many thanks Chris -- Chris Blunt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring calls into MeetMe
Hi All, I posted earlier with regards to three way calls and X-Lite, this kind of yielded everything I already suspected. However I suspect someone has a good working config for connecting a third party to an existing call (a-la-skype), or a detailed solution of using MeetMe to achieve this, without having to make two calls, transfer them in, then connect my self. Any help or insight really appreciated. Best regards Chris Blunt -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Cards connecting to BT
Hi, There are several people on the UK mailing list (I am one) that have purchased the TDM400P FXO and are having problems with disconnect. Basically the cards are great (sound quality etc) but give some issues with detecting a UK remote hang-up. Mainly an issue within IVR, MeetMe and VM. There are several of us trying to get to the bottom of this, either with fixes or workarounds. If you only want a couple of lines and ISDN isn't an option perhaps look at the Sipura 3000 they have one FXO and one FXS interface. Also they don't cost the earth are UK approved, and available in the UK so no import duty. Regards, Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: 14 February 2005 13:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Digium Cards connecting to BT Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXO - Any one got it working well in UK without Hangup problems
Hi Guys, I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone. Installation went great and call clarity is excellent no echo like I had on the clone card. My problems start with detecting hanging up the line. If a person calls into the system and speaks to me on a SIP phone when I hang up the call clears down OK, if the caller goes into an IVR, and hangs up a default timeout does a hang up and clears down OK. However if the incoming caller goes into MeetMe, and hangs up Asterisk doesnt detect this and sits there playing MOH indefinitely. The upshot is Asterisk and the TDM400P are not detecting a remote call hang-up. I have spoken to BT about what kind of disconnection etc and they are less than helpful. I have tried both busydetect and callprogress, (either or) and still no go. Is there anyone in the UK that has a working solution??? Any help appreciated before another sleepless night J Regards, Chris PS: Sorry if you have seen this posted elsewhere before -- SIP: [EMAIL PROTECTED] (ext 5002) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] # Transfers.
Thanks to Bruce for adding this stuff on attended transfers to the WIKI pages. I've been trying to get my head round this for a couple of days. Unfortunately I'm still having a bit of trouble. I have the latest CVS-HEAD, just downloaded and compiled. Added the bit for attended transfer into the Features.conf, and reloaded. However my phones just seem to ignore this. Do I need to change any other configs? Thanks Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended call transfer
Hi All, Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Potentially using a mix of phones would create confusion in a user base, any ideas on attended transfer or how to achieve this / mods to dial plan etc would be greatly appreciated. I have been on an almost vertical learning curve with Asterisk and Linux for 6 months this is just about my last challenge (for now haha). Many thanks Chris Blunt -- SIP: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with demo on asterisk
I'm by no means an asterisk Guru, just trying to get is together my self. How ever, no sound issues usually relate to blocked ports on your router / firewall. If your extension 1000 is an IAX connection, check your rtp.conf, and perhaps narrow the port range, allow port forwarding on this range (UDP) and port 5060 to your asterisk server. This seemed to do the trick for me. Hope this is of some use. Regards Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 18 January 2005 14:22 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with demo on asterisk Hi I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1 The process of installation was the following: First I compiled and installed Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the ztdummy (modprobe ztdummy) and then i installed Asterisk: make make install make configuration make samples I started Asterisk, and created one SIP account, with the following settings: sip.conf: [sipphone-1] type=friend host=dynamic dtmfmode=inband username=sipphone-1 secret=blablabla extensions.conf exten = 100,1,dial(SIP/sipphone-1) then I issued a reload on the asterisk command console I am using X-lite as SIP softphone. I configured the SIP proxy as given on the instructions on the site http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite I dialed the 1000 extension, and got connected, but there is no sound. I know that i should hear the demo comunication, but there is no sound. What am i doing wrong? Any help is welcome Regards Bozhidar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream BT100 - Asterisk - Voipjet ..... No ring ring when making a call
Hi All, Im sure this is something simple that I have missed somewhere. When I make a call using BT100 over IAX2 with Voipjet terminating I dont get a ringing sound whilst Im waiting to be connected. The destination party can answer the call (they do get ringing) and conversation can take place. I dont get this problem with X-Lite softphone? Any help appreciated and a very merry Christmas to all the * people out there Chris PS: Voipjet rocks! -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo - UK Impedance problem with X100P?
Hi, I have an X100p interface (clone). The system works fine but I get echo to a level where the system is all but unusable for IP PSTN. I seem to remember reading somewhere that the UK line impedance is different from the default compile and needs changing. I have Wikied etc, but found nothing yet. Any pointers appreciated. Regards Chris Blunt -- SIP: [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
RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?
Thank you to all that have posted so far. I realize the X100p clones are designed as voice modems. But if they are designed for the UK market and are BABT / EU approved, should they not support UK impedance? If these clone cards were capable of multiple impedance settings, how do we change Asterisk to take advantage of this (assuming we would have to make a change anyway for use with a TDM04B). Coincidentally I heard a rumour that the TDMxxx are not approves for use in the UK. Is this the case? Thanks again... Chris I have an X100p interface (clone). The system works fine but I get echo to a level where the system is all but unusable for IP PSTN. I seem to remember reading somewhere that the UK line impedance is different from the default compile and needs changing. I have Wikied etc, but found nothing yet. The x100p (and presumably the clones) have an integrated circuit on the board that was manufactured for use in the US with 600 ohm pstn lines. The chip cannot be changed to any other impedance. However, there can be many different sources for the echo and impedance matching is only one of them. Others include: - incorrect * zapata.conf parameters - poorly engineered motherboards (eg, poor PCI bus, interrupt latency) For zapata.conf, try something like: echocancel=yes echotraining=800 For poorly designed motherboards, there is no consolidated list of which ones are good/bad so you're left with trying another one on your own to see if it impacts the echo. The TDM04B digium card (as an example) does not use that same pstn chip, but rather another one from the same chip manufacturer. That chip does have support for something like 18 different country telco standards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?
Hi Soren, Thanks for your reply on this. My card is a clone, with an Ambiant 3200 chip. The parameter you gave me has sorted out many of my problems. It is people such as your self who are incredibly helpful within the Asterisk community. As like many others, I am relatively new to Asterisk and appreciate help in getting my proof of concept system up and running. Thanks again Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 12 November 2004 17:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo - UK Impedance problem with X100P? Rich Adamson wrote: [snip] For those that would like to play around with the above, might take a look at zaptel/fxstest.c (and the associated Makefile complile options) as it can be used to modify/view the tdm04b chip parameters. I'm not a programmer, but doubt whether it would take much to modify it to exercise the cards noted above. [snip] UK is a CTR21 country and after having a closer look at the wcfxo.c code it is supported *if* the card have the global chipset (Clone only, I believe). To enable CTR21 you have to modprobe/loadmod/whatever the wcfxo driver with the parameter 'opermode=1'. /Soren ___ 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] Anyone using Asterisk on Slackware 9?
Hi, I am trying to do a very minimal install of Slackware to run Asterisk on. Can anyone give me a list of what packages I need to install as I dont want X an all the associated bloat? Thanks in advance Chris -- ___ 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] Inbound Free World Dialup - extension not ringing?
I have opened up all the ports specified other than the 1-2 range as
my router just can't cope with that. Unfortunately I still get no sound.
Is IAX the best route or is registering my FWD connection through SIP.conf
the best solution, what do people recommend?
I have dug through the WIKI, and what instruction I can find, but to no
avail, examples or working confs would be fantastic just for comparison.
Thanks
Again
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward Eastman
Sent: 16 August 2004 00:44
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
IAX2 uses udp port 4569, so youll probably have to open that up on your
firewall/router.
http://www.voip-info.org/ is a good starting place for any asterisk problems
- specifically:
http://www.voip-info.org/wiki-Asterisk+firewall+rules
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
HTH
Ed
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt
Sent: 15 August 2004 23:06
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
Hi Lyle,
Thank you so much for your help, I think your information points to using
IAX2 rather than registering with FWD from the sip.conf
I have made an attempt to understand this, added the appropriate information
into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX
registration box, and I now get my local sip phone ringing when I dial in
from FWD! Hurrah, unfortunately I get no sound in either direction. Do
you have any experience of this or could it be due to me being inside a NAT
firewall? I have port 5060 forwarded to my * server, should I forward any
other ports? (I can only forward a maximum 20 single ports due to a
limitation on my home router).
As yet I am unable to make outgoing calls over FWD, I figured I would look
at this next.
Is there a NAT solution that could be used with sip.conf rather than the
IAX?
Again your help is most appreciated.
Best regards
Chris
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: 15 August 2004 15:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
You need a defination for the inbound FWD and what to do with that.
In my extensions.conf, I have:
[globals]
FWDNUMBER=123456 ; your actual fwd number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010
[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME} ; replace 123 with the desired
access code to dial out via FWD
exten =
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60
,r)
exten = _123.,3,Congestion
[local]
include = fwd_out :add to local context
[default]
;inbound dialing from FWD
exten = ${FWDNUMBER},1,Goto(housemenu,s,1) ; I have mine set to hit a
menu, no reason you cann't forward to an extension instead
- Original Message -
From: Chris Blunt
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 3:29 AM
Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?
Hi to all the * people out there,
Please kind to me as I am both new to Asterisk and to Linux But I am
learning fast.
My config is quite simple, Im just following examples and the Wiki: I have
two PCs running X-Lite phones, these work without problems between each
other, and I have a GS BudgeTone-100 registered to Free World Dial UP
(working no problem).
I have tried to set up Asterisk to accept calls from FWD on another number I
have registered, but I cant get my local X-Lite to ring on an inbound call
from FWD, and I get the busy tone on the BT100
When I sip debug, I can see that I am registered with FWD, and when I call
the number from the BT100 I can see all the incoming information but still
nothing on my X-Lite.
My extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten = 1,1,Dial(SIP/phone1,20,tr)
exten = 2,1,Dial(SIP/phone2,20,tr)
exten = 2,2,VoiceMail,u1234
exten = 2,102,VoiceMail,b1234
;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain,s1234
exten = 6601,1,WaitMusicOnHold(60)
exten = 232999,1,Dial(SIP/phone1,30,tr)
exten = 232999,2,Hangup
I am behind a NATed fire wall, but Im not sure that is related.
Any ideas or help (working simple confs) would be much appreciated.
Best regards
--
Chris Blunt
SIP: [EMAIL PROTECTED]
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[Asterisk-Users] Inbound Free World Dialup - extension not ringing?
Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux But I am learning fast. My config is quite simple, Im just following examples and the Wiki: I have two PCs running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to set up Asterisk to accept calls from FWD on another number I have registered, but I cant get my local X-Lite to ring on an inbound call from FWD, and I get the busy tone on the BT100 When I sip debug, I can see that I am registered with FWD, and when I call the number from the BT100 I can see all the incoming information but still nothing on my X-Lite. My extensions.conf: [general] static=yes writeprotect=no [globals] [sip] exten = 1,1,Dial(SIP/phone1,20,tr) exten = 2,1,Dial(SIP/phone2,20,tr) exten = 2,2,VoiceMail,u1234 exten = 2,102,VoiceMail,b1234 ;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr) exten = 1001,1,Ringing exten = 1001,2,Wait(2) exten = 1001,3,VoicemailMain,s1234 exten = 6601,1,WaitMusicOnHold(60) exten = 232999,1,Dial(SIP/phone1,30,tr) exten = 232999,2,Hangup I am behind a NATed fire wall, but Im not sure that is related. Any ideas or help (working simple confs) would be much appreciated. Best regards -- Chris Blunt SIP: [EMAIL PROTECTED]
