[asterisk-users] AMI Redirect from in a WaitExten
Hi, I have and AMI application that tries to redirect a channel if a certain condition exists. It seemed to work when using Asterisk version 11.14, but now I am trying it with 11.19 and it is not. Here is the scenario: 1. A channel connects to the dialplan and is put into a WaitExten () 2. The channel gets some DTMF codes (the AMI app also captures the DTMF string as it is typed) 3. The AMI app recognizes the DTMF string as a command for it to redirect the channel to a special exten which handles some special functions (the DTMF string is put in a channel variable with a Setvar action before the Redirect action) The problem seems to be that when the AMI Redirect action is sent the WaitExten is redirecting to whatever DTMF string has already been entered, not the Exten that the Redirect action specified. Even thought the Exten does not exist it the dialplan. If I stop the AMI app the WaitExten app does not redirect to the dialed DTMF (it should not since it does not match any Exten in the dialplan). If I replace the WaitExten with a simple wait it will work properly, but this will not allow for regular extension to be dialed. Example: 1. User dials 00# 2. AMI attempts to Redirect to ## 3. Asterisk log show executing 00# even though 00# does not exist anywhere in the dialplan Is there anyway to change the behaviour of the WaitExten application to let the AMI app Redirect a channel from it? Is this something that changed (intentionally or accidentally) from 11.14 to 11.19 or I am missing something? Thanks in advance, Neil Cherry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI Redirect both calls from a bridge
On 2014-12-17 9:34 AM, Karsten Wemheuer wrote: Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. I think You have to do it in two steps. First connect both legs with a conference and then connect each one with the final extension. You didn't tell, which version of asterisk You are using. In 11 and later there is the new conference module, which makes it easier. In the first step You can use AMI REDIRECT to transfer both parties into one dynamic conference. Use the Channel and ExtraChannel to take both channels. In the second step use AMI Join Events to trigger your next transfer to the different extensions in Your dialplan. Each channel joining the conference will generate a separate event. HTH, Karsten Thank you for your response. I am using Asterisk 11. My issue was simpler than your solution, in that I was simply able to use the ExtraChannel on the Redirect command to send the second channel directly to the second exten and not have it dropped. My problem was that I did not realize that I could use the "Extra" fields in the redirect command. The scenario was while two channels are in a bridge (call) I want to put one on hold and supply dial tone to the other. I have an exten toredirect the on hold channel to and a dial tone exten to redirect the other channel to, but I was trying to do each in a separate redirect instead of combining them in one redirect action. Thanks for you help, Neil Cherry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI Redirect both calls from a bridge
On 2014-12-17 9:08 AM, Neil Cherry wrote: Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. Thanks, Neil Cherry Please disregard for now, I have found some documentation the refers to an extra channel on the Redirect action. I will try that. Thanks, Neil Cherry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI Redirect both calls from a bridge
Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. Thanks, Neil Cherry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ADSI Scripting Documentation
Hello, I know this is an out of date technology, but I have a need to use it to reprogram some old phones. I have managed to get Asterisk and a Digium board to upload scripts to the phones using the asterisk.adsi example, but trying to make my own scripts based on the example file is proving very limiting and difficult. I am sure there are a lot of features I am missing since I do not have documentation on the scripting language. Does any one have any old documentation or know where I can get any ? I found references in some list archives to this link: http://publibfp.boulder.ibm.com/epubs/pdf/wvrpad04.pdf But it has been remove from the IBM site. If anybody has a copy of this PDF file and can send me a copy it would be really appreciated. Thank, Neil Cherry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL & Blacklist question
Does the ${BLACKLIST()} function allow for values other than 1 to be
returned and if so how can I use that is the AEL? Can I use the
function in a switch statement?
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[asterisk-users] How do I debug?
I had my setup working properly under 1.2 and after a disk crash I decided that I wanted to try Asterisk 1.4. So far I can transfer between phones and I can dial out. What I can't get working is to get an SPA-3102 to 'send the calls' to Asterisk. I have the device added to the sip.conf file and it shows up in users and peers (but not in registry). Where do I start with debugging? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog Author of: Linux Smart Homes For Dummies ___ --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] Survey: In what ways do you use Asterisk at your house?
Earle Clubb wrote: - What service provider/technology do you use for origination/termination? - What hardware/software do you use and how does it all tie together? - What tasks do you use * to accomplish? - Any other pertinent info. Until last summer I had Asterisk doing the normal call handling my home. You know selecting which line to call out on via an SPA-3000 and SPA-3102. We do have trouble with the SPA's as the echo can be quite bad or the volume is quite low (take your pick). I'm also routing various calls to various vm-boxes and sending selected callers to the SIT. I also had an extension that interfaced to Mr. House home automation software. I could control and monitor a few things in my home. This system is no longer working due to a drive crash and the lack of backup for parts of this setup. I'm hoping to get the time towards the end of the year to put it back together. I may try to integrate the voice recognition (Sphinx) into the setup also. This was running on a 1GHz/512M/300G vanilla x86 clone. I had printer services, DNS, DHCP, file sharing, home automation, Asterisk and a few other things running. It's also my development system. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, Linksys SPA-3000 echo
I've found that when I use an IP VoIP device (phone or ATA) to the SPA-3000/3102 (I have both) that I have the echo problems. While I've learned to deal with it my wife won't. I have no such problem with the POTS phone directly connected to the SPA. I have one SPA - ATA - VoIP service and an SPA to POTS line (~2 cable miles from the CO). I'm pretty sure my problems are software echo cancelers doing battle with each other but I have no proof. BTW, how do you turn off the software echo canceling in the SPA? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --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] F3000 registering to asterisk
Paul Hayes wrote: Neil Cherry wrote: [snip] How did you get access to the web config? What user and is it the default password/access code? type it's IP address into a web browser. Username: admin, password: psw is the default. Thanks, I was missing the psw. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --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] F3000 registering to asterisk
Matt wrote: Hi, I have an F3000 phone that I am trying to register to asterisk. As far as I can tell I have everything in correct. Are there any little quirks I need to worry about? The phone has internet access, set it's time.. I can access the web config, but it just won't register with asterisk. I don't see anything meaningful in the full log. I have my phone working with my Asterisk (I do have various issues). From sip.conf: [2215] username = 2215 secret= 2215 type = friend host = dynamic port = 5060 context = from-internal callerid = "Livingroom WiFi" <2215> mailbox = 2215 nat = never dtmfmode = rfc2833 canreinvite = yes ; qualify = yes insecure = very disallow = all; need disallow before we can allow allow = ulaw I can't get it to qualify but I think that's a problem with my wireless access point. How did you get access to the web config? What user and is it the default password/access code? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --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] Canreinvite
Il Neofita wrote: I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? Are you using the same codecs on the SPA3000 and the xlite? If no then there's your reason. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --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] DHCP configuration for Cisco 7960?
Julian Dunn wrote:
(Apologies to those Toronto Asterisk Users' Group folks who have seen
this message... I figured I'd have more success with a wider audience)
I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on
FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on
the wire. I wonder if anyone has done this before and therefore can
validate whether or not the traffic I am seeing is normal.
I have dhcpd.conf set up like this
---8<--- cut here ---8<---
option cisco-etherboot-server code 150 = ip-address; .
.
.
host c7960 {
hardware ethernet 00:16:46:9B:6D:62;
fixed-address 192.168.5.14;
option host-name "c7960.acf.aquezada.com";
option cisco-etherboot-server 192.168.5.7; }
host c7960 {
hardware ethernet 00:0d:11:22:33:44;
fixed-address 192.168.2.196;
option domain-name "uucp" ;
option tftp-server-name"192.168.2.1";
always-reply-rfc1048 true ;
}
Here's mine, works under Linux.
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[Asterisk-Users] Asterisk - SPA-3000, 407 error
I recently lost my setup (bad drive) and I'm now trying to get my setup back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an SPA-3000. I can call the phone extension, I can call from the phone on the SPA to other extensions and I can call out to the PSTN. What I can't do is to call from the PSTN to through the SPA to Asterisk. It rings twice then I get fast busy. I have the SPA wait for the caller ID info (the reason it rings twice). What I've got is as follows: Code: registry=>pstn:[EMAIL PROTECTED]:5061 [pstn_in] username = pstn secret= pstn type = user host = spa.uucp port = 5061 context = from-pstn mailbox = 2202 nat = never dtmfmode = rfc2833 canreinvite = yes qualify = yes insecure = very disallow = all; need disallow before we can allow allow = ulaw ; [pstn_out] username = pstn secret= pstn type = peer host = spa.uucp port = 5061 context = to-pstn from_user = pstn nat = never dtmfmode = rfc2833 canreinvite = yes qualify = yes insecure = yes disallow = all; need disallow before we can allow allow = ulaw ; My dial plan on the SPA looks like this: (S0<:[EMAIL PROTECTED]>) What I see on the Asterisk console is : May 17 23:36:29 NOTICE[11306]: chan_sip.c:10326 handle_request_invite: Failed to authenticate user xxx ;tag=blahblahblah In sniffer I traces I see a 407 (Proxy Authentication Required). I've also noticed a 401 (Unauthorized) for the phone on the SPA. What am I configuring incorrectly. I'm using Asterisk 1.2.7, previously I was using 1.2.0. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --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] Can someone explain the 's' extension
Eric "ManxPower" Wieling wrote: Neil Cherry wrote: Funny thing is I can get Asterisk to use the 's' extension as a catch all (I use the include => xcontext command). But I needed to describe it properly for chapter in a book I'm writing. Man I hope I get this stuff right! BTW, I'm using SIP extensions to do all my testing. Works great (when I remember to include the correct contexts ;-) exten => s is NOT a "catchall" it's more of a "catch nothing" i.e. it only catches calls that have no destination info. A "catchall" would be exten => _. but that would catch extensions that are not numbers (like o, i, t, T, h, etc). A catch all number extensions would be something like exten => _X. Thanks, you just pointed out 2 huge mistakes on my part. The first is what I thought 's' was for and the second is that my above statement is untrue. I used the '_.' as the catch all. I'll correct that. Again thanks! -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Can someone explain the 's' extension
Eric "ManxPower" Wieling wrote: Neil Cherry wrote: Does someone explain the 's' extension? In the Wiki it says it's the catch all extension. In the Asterisk 1.2-rc1 it say it isn't but doesn't say anything more. Needless to say I'm confused. When a call comes into Asterisk (PSTN, VoIP, etc) and call has NO information as to what extension to route to then Asterisk will try sending the call to extension => s In practice this only happens if you have a voice T-1 (Not PRI) with no DIDs, or if you have an analog FXO port. Wow, thanks guys for the quick response. I'm checking out the pdf file (previous list message). Funny thing is I can get Asterisk to use the 's' extension as a catch all (I use the include => xcontext command). But I needed to describe it properly for chapter in a book I'm writing. Man I hope I get this stuff right! BTW, I'm using SIP extensions to do all my testing. Works great (when I remember to include the correct contexts ;-) -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Can someone explain the 's' extension
Does someone explain the 's' extension? In the Wiki it says it's the catch all extension. In the Asterisk 1.2-rc1 it say it isn't but doesn't say anything more. Needless to say I'm confused. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] DSL router with QOS
Rod Bacon wrote: What's the point? You can prioritise as much as you like at your own end, but as soon as it leaves your premises and enters the 'net, all bets are off! While I do this for a living (networks and QoS) and I shouldn't argue with the above, I will. The engineer in me just has to have the egress policed. :-) Even the contention ratio of the DSL circuit (as provided by your ISP) can kill you. QOS is really only useful in a point-to-point scenario, or in a meshed network that honors QOS on all links. If you really want to experiment, grab an old PIII for $50 off e-bay, and setup a linux box as a router behind your DSL modem. You can play with QOS as much as you like then, without forking out $500. Rod makes several really good points here. If you want a box with Wireless and Traffic shaping get the Linksys and use one of the 3rd party firmware images. I'm currently using a WRT54G with Sveasoft's Alchemy image, though I now heavily disagree with the vendor's opinion of Open Source and it's community (I won't renew my subscription). My understanding is that the OpenWRT and another image may also have this same functionality (Traffic Shaping called tc). This may be a good way to go and the cost should be less than $100 (US). You'll still need the DSL modem but I don't like all in one devices at that level. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Cisco 7940 - TFTP
Kris Edwards wrote: Hey all. I recently got the above mentioned phone and am having trouble upgrading the firmware. I have the sip firmware (availabe but not loaded), but wouldn't mind using sccp it's just that I can't even get the config file to load via tftp. The phone loops requesting CTLSEP.tlv then will, for whatever reason start asking for the SEP.cnf.xml file even though the .tlv file was never sent. I can see the requests coming onto my gentoo box, but my machine never replies, so I'm not sure if it's an issue with my tftpserver or just this (used) phone. I've read several articles on the wiki, in particular, this one http://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960 , discussing this problem. So, here are a couple of questions. 1) I'm using a null file for CTLSEP.tlv.. is that sufficient, or does the file actually need something (empty set of xml tags or something) 2) Is there any other way to get the files on the phone?? (even w/ tftp, can I put them there? I scanned the phone and the only listen port i see is 80) I'm using netkit-tftp server (atftp crashes for some reason) and I've chmod 777 the /tftproot dir. No firewall between phone and server. I'm not home at the moment, but will be shortly so if anyone is interested in helping (or knows of a link that might help [i think i've exhausted the wiki]), let me know what info you need and I will provide it. This paperweight looks like it would make a cool phone for me :) First, my tftp server is a Linux box and my tftp directory is /tftpboot (setting can be found in /etc/xinetd.d/tftp if your using xinetd such as Fedora is). Next, I get requests from the 7960 (similar phone) running the SIP image (I've tried v5, v6 and I'm now on 7.4). I get the following files requested when the phone boots up: CTLSEP000ABCD01234.tlv SEP000ABCD01234.cnf.xml SIP000ABCD01234.cnf MGC000ABCD01234.cnf XMLDefault.cnf.xml SIPDefault.cnf SIP000ABCD01234.cnf P0S3-07-4-00.loads P0S3-07-4-00.sb2 RINGLIST.DAT dialplan.xml Note that 000ABCD01234 is a made up MAC address, replace it with your phone's MAC address. It also requests the files listed in the RINGLIST.DAT. Take a look at: http://www.loligo.com/asterisk/Cisco/79xx They have lots of info (you'll still have to do some work). Sorry I can't give credit to the person who create all this but I'm sure they're on this list (that's where I found this info). -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial Plan
Felix Amaral wrote:
The Asterisk I biult only does outbound calls, and it do them by LAN, I
don´t have any special hardware. Please help with the Dial Plan.
I don't know maybe I'm being a bit snippy (a bit upset) but you don't
seem to be making any effort to do any work. My example is not a
great example but it's not bad either. Try removing the 'SIP/pstn'
portion and putting in what you need (since you aren't telling us
I'm not guessing either). You need to do a little home work also.
I know there are examples out there for just about every service.
I found them for plenty when I did a Google search for my home
setup.
exten => _1.,1,Dial(SIP/pstn/${EXTEN}) ; Unanswered, Do we ever get here?
exten => _1.,2,Playback(allison7/all-circuits-busy-now)
exten => _1.,3,Playback(allison7/pls-try-call-later)
exten => _1.,4,Macro(hangupcall)
; Busy, Or do we always go here?
exten => _1.,102,Playback(allison7/pls-try-call-later)
exten => _1.,103,Playback(allison7/all-circuits-busy-now)
exten => _1.,104,Macro(hangupcall)
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Re: [Asterisk-Users] Dial Plan
Felix Amaral wrote:
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and made it
work between diferrent extensions in the office and now I need to make it
work on calling outside the office and I think I need a Dial Plan, can
somebody help me a little with this?
I have the following at the bottom of my context for my SIP
extensions:
exten => _1.,1,Dial(SIP/pstn/${EXTEN})
; Unanswered, Do we ever get here?
exten => _1.,2,Playback(allison7/all-circuits-busy-now)
exten => _1.,3,Playback(allison7/pls-try-call-later)
exten => _1.,4,Macro(hangupcall)
; Busy, Or do we always go here?
exten => _1.,102,Playback(allison7/pls-try-call-later)
exten => _1.,103,Playback(allison7/all-circuits-busy-now)
exten => _1.,104,Macro(hangupcall)
I only have a Grandstream BT100 & a Sipura SPA-3000 (named line1 &
pstn). The SPA-3000 registers pstn with asterisk and is able to
handle all the dial outs. Since I no longer have 7 digit dialing
(we have overlays which require dialing at least 10 digits). So
any number that starts with 1 will be sent out to the PSTN. I
haven't setup the emergency services number or other numbers
such as emergency dialing. I really aught to do that in case
someone accidentally picks up my extra test phones.
BTW, this is for home use.
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Re: [Asterisk-Users] Dial plan questions
Neil Cherry wrote: Humberto Aicardi wrote: Neil, When you use the Dial command you must specify the device to use for dialing, so you cannot use Dial(2201,20) you must use Dial(SIP/2201,20) which informs to use the the SIP device 2201. Ah, thanks! That worked with a second extension I set up (SIP/2202). It goes to busy on the SIP/2201 (I test ring, busy & unavailable on SIP/2202 and it works fine). So now I need to figure out why Asterisk sees SIP/2201 as (Unspecified). mozart*CLI> sip show peers Name/username HostDyn Nat ACL Mask PortStatus 2202/2202 192.168.24.197 D 255.255.255.255 5060OK (16 ms) 2201/2201 (Unspecified)D 255.255.255.255 0 UNKNOWN pstn/pstn 192.168.24.197 D 255.255.255.255 5061OK (16 ms) 3 sip peers [2 online , 1 offline] Dang Grandstream! Fixed the (Unspecified) problem by doing a SIP reload. Weird. Dang Asterisk? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial plan questions
Humberto Aicardi wrote: Neil, When you use the Dial command you must specify the device to use for dialing, so you cannot use Dial(2201,20) you must use Dial(SIP/2201,20) which informs to use the the SIP device 2201. Ah, thanks! That worked with a second extension I set up (SIP/2202). It goes to busy on the SIP/2201 (I test ring, busy & unavailable on SIP/2202 and it works fine). So now I need to figure out why Asterisk sees SIP/2201 as (Unspecified). mozart*CLI> sip show peers Name/username HostDyn Nat ACL Mask PortStatus 2202/2202 192.168.24.197 D 255.255.255.255 5060OK (16 ms) 2201/2201 (Unspecified)D 255.255.255.255 0 UNKNOWN pstn/pstn 192.168.24.197 D 255.255.255.255 5061OK (16 ms) 3 sip peers [2 online , 1 offline] Dang Grandstream! -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial plan questions
I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/[EMAIL PROTECTED],20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as unavailable. This one does not work: exten => 2401,1,Dial(2201,20,) exten => 2401,2,Voicemail(u2201) exten => 2401,3,Hangup exten => 2401,102,voicemail(b2201) exten => 2401,103,hangup If I dial 2401 I get fast busy, what am I doing wrong? mozart*CLI> sip show peers Name/username HostDyn Nat ACL Mask PortStatus 2202/2202 192.168.24.197 D 255.255.255.255 5060OK (17 ms) 2201/2201 (Unspecified)D 255.255.255.255 0 UNKNOWN pstn/pstn 192.168.24.197 D 255.255.255.255 5061 OK (15 ms) -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk on windows
Matt wrote: Why on earth would you want to run it on Windows? First off, your performance is going to go down because of the GUI... oh your call quality just went down the toilet? Yeah sorry the screen saver just kicked in. Having issues making calls? Oh sorry we had to reboot for a critical update. Yeah I know audio isn't working right, the swap file is a little large right now, we need to reboot. Are you on crack?!?! Asterisk runs well on Linux because of the lack of a GUI... sleek simple interface (text) to it. Linux is free, windows adds a license cost. Since you shouldn't be running any other applications on the server anyway, why not just install Linux? Trying to run it on windows seems like a bad idea to me. I think this poor user saw the 2.0 announcement and thought it was real. Somebody should change that to 13.0 instead, I nearly freaked when I saw it until I realized it was a joke. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] BT100 can't register
My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call out from my BT100 to other extensions and through the SPA to the POTS line. Don't assume I really know what I'm doing. One minute it all makes sense and the next I'm clueless. Thanks Oh, I trimmed the sniffer trace to only include the SIP decode. == [2201] username = 2201 authuser = 2201 secret= 2201 type = friend host = gs1.uucp ; host = 192.168.24.192 port = 5060 context = from-internal callerid = "Grandstream" <2201> mailbox = 2201 nat = never dtmfmode = rfc2833 canreinvite = yes qualify = yes ; qualify = no ; outgoinglimit = 2 ; permit only 1 outgoing call at a time ; incominglimit = 1 ; disable callwaiting signal (2nd call to phone) disallow = all ; need to disallow=all before we can use allow= allow = ulaw; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow = alaw == No. TimeSourceDestination Protocol Info 27 453.810961 gs1.uucp mozart.uucp SIP Request: REGISTER sip:asterisk.uucp(remove all bindings) == Session Initiation Protocol Request-Line: REGISTER sip:asterisk.uucp SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.24.192;branch=z9hG4bKd57bf9e7269cdaee From: "Neil J. Cherry" ;tag=b946eeaaed68b378 SIP Display info: "Neil J. Cherry" SIP from address: sip:[EMAIL PROTECTED] SIP tag: b946eeaaed68b378 To: SIP to address: sip:[EMAIL PROTECTED] Contact: * Supported: replaces Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream BT110 1.0.7.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 == No. TimeSourceDestination Protocol Info 28 453.811410 mozart.uucp gs1.uucp SIP Status: 403 Forbidden(1 bindings) == Session Initiation Protocol Status-Line: SIP/2.0 403 Forbidden Status-Code: 403 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.24.192;branch=z9hG4bKd57bf9e7269cdaee From: "Neil J. Cherry" ;tag=b946eeaaed68b378 SIP Display info: "Neil J. Cherry" SIP from address: sip:[EMAIL PROTECTED] SIP tag: b946eeaaed68b378 To: ;tag=as6453730d SIP to address: sip:[EMAIL PROTECTED] SIP tag: as6453730d Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 ====== -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo on SPA-3000 FXO
Chris Mason (Lists) wrote: After rebooting the SPA-3000, the internal users of calls routed through the FXO interface hear pretty bad echo. This persists for days, maybe more than a week. At some point, the echo goes away. This is a known problem with the spa3k-3.1.5b firmware, I have made them aware of it and hopefully they are working on a fix. I'm also seeing this with Software Version: 2.0.13(GWg) Hardware Version: 2.0.1(7567) I also have the SPA to PSTN gain down to -3 and PSTN to SPA to +3 (I've been having trouble hearing the people I've called). The echo was so bad at one point I was hearing multiple echoes. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] [OT] SPA-3000 loudness
Just a note to those of us with the SPA-3000 that have had a hard time with the 'volume' of calls from the PSTN (we're on the line they're on the PSTN). Anyway I change the parameter: PSTN To SPA Gain to 6 and the volume is fine (it's under the PSTN Line tab). I don't remember seeing this on the list so I thought I'd add it here. Guess I should add it to the wiki (a little later). -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] No dial tone on BT100
I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 - Asterisk is Asterisk asking the BT100 for it's option (102 Options) and the BT100 not replying. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] IAX2, can't receive calls
Wilson Pickett wrote: I have IAX2 (FWD) partially working. I can place calls from my Asterisk box but I cam unable to receive them (comes back as busy). I have my firewall forwarding the udp ports 5060, 4569, 5036 and 1 thru 2 to my asterisk server. I think I have the firewall correctly setup as I can forward other services to their appropriate servers. I have no mail box on the one account (the one I'm testing to). I've followed the FWD instructions but I've had no luck. what does iax2 show register and iax2 show peers show wrt FWD? mozart*CLI> iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569xx 69.142.122.219:456960 Request Sent mozart*CLI> iax2 show peers Name/UsernameHost Mask Port Status fwd2/xx 69.90.155.70(S) 255.255.255.255 4569 Unmonitored 1 iax2 peers [0 online, 0 offline, 1 unmonitored] Hmm, I do seem to have a problem but this is not what caused my post (but I do have to fix that). I seem to be loosing registration for about 60 seconds or so (it's registered, it's unregistered). You're not going to beleive this, it turns out the problem is with my BT101 and/or Asterisk config related to the 101. I thought I had the 101 properly configured but when I switched to extenstion 2210 (a different extension) it started working. For some reason the BT101 isn't registered (seems none of my SIP phones are). Let me do some more work and I'll post a new message under the proper heading for that problem. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] IAX2, can't receive calls
I have IAX2 (FWD) partially working. I can place calls from my Asterisk box but I cam unable to receive them (comes back as busy). I have my firewall forwarding the udp ports 5060, 4569, 5036 and 1 thru 2 to my asterisk server. I think I have the firewall correctly setup as I can forward other services to their appropriate servers. I have no mail box on the one account (the one I'm testing to). I've followed the FWD instructions but I've had no luck. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] 7960 SIP Firmware Upgrade Strange Problem
Walid Azab wrote: Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Walid, you should be able to unzip it anywhere and copy it into the directory. It sounds like a permissions problem when you copied it. In the future just make sure that files copied into the tftp directory have at least read permission for everyone (user, group and other). Since it's working now you don't need to fool with it. Just information for the future. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Help with Cron and Reload
Federico Alves wrote: This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom [EMAIL PROTECTED] Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: [EMAIL PROTECTED] (Cron Daemon) To: [EMAIL PROTECTED] Subject: Cron <[EMAIL PROTECTED]> asterisk -r -x reload X-Cron-Env: X-Cron-Env: X-Cron-Env: X-Cron-Env: /bin/sh: line 1: asterisk: command not found Any ideas? Login in as root, type in: type asterisk I get /usr/sbin/asterisk Change crontab (crontab -e) to use the full path. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] What do you name yours
trixter http://www.0xdecafbad.com wrote: On Tue, 2005-05-10 at 23:46 -0400, Black Ratchet wrote: My two toy boxes at work are 'succasunna', named after the 1st city to ever have #1ESS phone switch, and 'murrayhill', which was named after the location of the AT&T headquarters. I wonder what that big building in basking ridge was then. Verizon HQ? ;-) To those that don't understand, it looks like Verizon has purchased the Basking Ridge building. I think it was initially sold to an insurance company but I'm not sure. And Murray Hill was the Bell Labs headquarters, I think. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] IP Phones for home use?
Thanks for all the opinions I have a lot of good examples now. I do like the 7940G and I may borrow one from the lab and see how well it goes. I'm also happy to have the rest of the opinions and I've not eliminated the other vendors just slimmed it down to fewer models. Lastly my wife informed me that she wants this really ugly phone to be put in the kitchen. Looks like a phone with a crank on the side. I know better than argue so I'll be getting one of the Sipura gateway devices. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] IP Phones for home use?
I currently have Asterisk, an X100P and a Grandstream 100 for playing around with. I'm interested in another IP phone for daily use. The Grandstream is fine but feels a little cheap and won't be acceptable by my wife (very important). What I want is a phone with lots of buttons and an LCD screen with lots of information (the Cisco 7970 would be nice)! But the reality is that we need a phone that is not overly complicated, handles 2 calls, caller ID info, reasonably priced, not ugly looking (the 7920's are ugly) and not much else. So a compromise to something in between is called for. What are your recommendations for a slightly fancy home phone? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Is there hardware to remote control
Lyle Giese wrote: I found this interesting box at qkits.com QK108 It has 8 relay outputs and 4 inputs. It's controlled via RS-232c(serial port) instead of a printer port. I have an 8 port serial card in a linux server to control a bunch of stuff. I have apache on that server and can control the relays via a cgi script. I found it very easy to program a serial port via perl and with an 8 port serial card(from Perle). You can have a bunch of stuff hanging off it, like a 4 probe temp kit (QK 145) and a ups and an RS-232 voltmeter to monitor the commerical power coming in and I suspose it would be easy to take this even further to write AGI scripts and dial an extension and let * announce the temperature or status of those inputs and to control the outputs of the QK108. I also use it with their K2639 to monitor the sump pits, monitoring for sump pump failure.(therefore high water levels). Cool, I've got to check that out. For the OP, check my 1st web page below for various odds and ends for the hardware and software that can be run on a *nix box. It's not directly * related but if you can run scripts you can take advantage of my collection of links. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Re: cannot call Grandstream
David Ishmael wrote: I think my Netgear router will try to lease the same DHCP address to a device based on MAC automatically each time the device queries for an address (but I'm not 100% sure about that, never really watched it). So the problem is with the address changing? I can't infer that from the 2 examples as it may be some other problem with the DHCP implementation on the DHCP server. Though it may be a possibility. I like to have the stationary IP devices to have a permanent IP address. It just makes it easier to admin my local DNS (I have too many devices to remember all the IP addresses). -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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: cannot call Grandstream
David Ishmael wrote:
I was considering a GS phone (102 or 102D), what version of the GS are you
using? Do all GS phones have issues with DHCP? I use DHCP on my network so
I want to make sure I understand potential issues before making any
purchases.
I have my GS101 working with DHCP, I setup my dhcp server to give
out the same address each time. Like this
host bt101a {
hardware ethernet 00:0b:82:xx:xx:xx;
fixed-address 192.168.2.192;
option routers 192.168.2.254;
option domain-name-servers 192.168.2.10;
option domain-name "uucp" ;
option log-servers 192.168.2.10 ;
# option time-servers192.168.2.10 ;
option time-offset -18000; # Eastern Standard Time
always-reply-rfc1048 true ;
}
I never could get my time server to work with the GS.
--
Linux Home Automation Neil Cherry [EMAIL PROTECTED]
http://home.comcast.net/~ncherry/ (Text only)
http://hcs.sourceforge.net/ (HCS II)
http://linuxha.blogspot.com/My HA Blog
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Re: [Asterisk-Users] Graceful CLI/crontab reboot
Marcelo Pacheco wrote: Em Qua 20 Out 2004 15:14, Andrew Edmond escreveu: Asterisk Community -- I'm looking for a way to gracefully shutdown asterisk at least once a day and bring it back online. I'm using Gentoo Linux and using safe_asterisk from /etc/init.d/asterisk. #!/bin/bash export PATH=/sbin:/usr/sbin:/usr/local/sbin:/usr/local/bin:/bin:/usr/bin /usr/sbin/asterisk -r -x "stop when convenient" rmmod wcusb rmmod wcfxo rmmod zaptel modprobe wcusb modprobe wcfxo ztcfg -vvv /usr/sbin/asterisk I do exactly what you want. But I also restart 100% all of zaptel. Needed for the USB FXS module. One thing to note about Gentoo is that it doesn't like when you stop a service any other way but the star-start scripts. Marco's idea is a good one but use the /etc/init.d/asterisk script to stop, clean up with the above and restart. If you do stop asterisk with/without the start script make sure to remove the /var/lib/init.d/started/asterisk link otherwise Gentoo won't let you restart the asterisk script (ARGH!). -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Graceful CLI/crontab reboot
Andrew Edmond wrote: Asterisk Community -- I'm looking for a way to gracefully shutdown asterisk at least once a day and bring it back online. I'm using Gentoo Linux and using safe_asterisk from /etc/init.d/asterisk. Anybody have a handy CLI/crontab script that accomplishes this? Andrew ___ 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 How about putting this in cron, as root type crontab -e and then add the following: # reload asterisk every day at midnight exactly $00 00 * * * /etc/init.d/asterisk stop ; sleep 2 ; /etc/init.d/asterisk start -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] IAX error messages
I just setup * to work with FWD and I'm now seeing these error messages: IAX Packet 31216 from circuit ids 212->1conflicts with earlier call with circuit ids 1->124 What can be causing this? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] New Open Source Project: Asterisk Management Portal
Jason Becker wrote: Details of the project can be found here: http://amp.voxbox.ca One word of warning backup your settings (/etc/asterisk/*) before installing it will overwrite your files. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Telco POTS -> FXO ?
I really found it now! I rewire the jack for 2 wires and I no longer get the tone problem. I think it was crossed wires. Now onto checking out Asterisk ... -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Telco POTS -> FXO ?
[EMAIL PROTECTED] wrote: On Wed, 13 Oct 2004, Neil Cherry wrote: OK, I found the problem! The card must be bad, maybe the pair is reversed. When I power down the server (power off, ac cable disconnected) I still get the tone! I've had enough of this I think I'll get another card maybe the TDM400P and an FXO to start with. ARGH! Complete flyer, but disconnect all other devices on the line. And: if this is a line with ADSL on it, make sure you have a filter. Complete flyer ??? No DSL but I'll try a few more lines, when I get the chance. BTW, the phone/pots line behaves well with no X100P on the line. Thanks -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Telco POTS -> FXO ?
Kanuri, Seshu (Company IT) wrote: Correct. Line as in "Wall Jack" not as in "Phone". You have to connect your FXO card with a RJ11 cable between your telephone wall socket and the RJ11 Port in the FXO card. OK good, I think I have an FXO card (the wcfxo module installs and ztcfg -vv gives no errors with this one). The reason I'm so confused is that without asterisk running, only wcfxo, zaptel and crc_ccitt insmod'd, I can pick up the pots line phone (on the phone jack and telco pots line on wall/line) and I hear tone (not the telco tone or at least tone riding on the telco tone). When I dial a phone (w/CID unit, not on asterisk) I get a busy tone and a ringing!. Eventually I get the answering machine and the busy signal. OK, I found the problem! The card must be bad, maybe the pair is reversed. When I power down the server (power off, ac cable disconnected) I still get the tone! I've had enough of this I think I'll get another card maybe the TDM400P and an FXO to start with. ARGH! -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Telco POTS -> FXO ?
Maybe I'm just doing this wrong. Is the FXO card (X100P) used to connect to the telco pots line? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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 sending out tone all the time?
Ilia Mirkin wrote: You want to use the wcfxo module with the X100P. wcfxs is for the TDM400P card. On Wed, 2004-10-13 at 03:43, Neil Cherry wrote: I'm in the process of setting up the X100P card and I am getting continuous tone on the X100P but only if plugged into the POTS line. Here is what I have so far: # lsmod Module Size Used by wcfxs 26912 0 zaptel223460 1 wcfxs crc_ccitt 1920 1 zaptel # cat /etc/zaptel.conf # loadzone = us defaultzone=us fxsks=1 Thanks! That's a little better. The error has gone away but I still get the tone on the line. I'm certain I'm plugged into the correct jack. # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. # lsmod Module Size Used by wcfxo 12064 0 sg 23708 0 zaptel223460 1 wcfxo crc_ccitt 1920 1 zaptel rtc10424 0 usbcore 108644 1 mxser 25948 0 via_rhine 17416 0 -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] X100P sending out tone all the time?
I'm in the process of setting up the X100P card and I am getting continuous tone on the X100P but only if plugged into the POTS line. Here is what I have so far: # lsmod Module Size Used by wcfxs 26912 0 zaptel223460 1 wcfxs crc_ccitt 1920 1 zaptel rtc10424 0 usbcore 108644 1 mxser 25948 0 via_rhine 17416 0 # cat /etc/zaptel.conf # loadzone = us defaultzone=us fxsks=1 # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Earlier I didn't get the above error. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Updated Grandstream configurator
Stephen R. Besch wrote: The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should have at least SP3, or even better, SP4 on Win2k. I believe it will run on Win9x, but I have not tested it and can make no guarantees. Thanks, I've been having real trouble with those stupid DLLs. I can't upgrade some of them no matter what I do (WIN2K)! -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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] FINALLY! a good book about Asterisk.
Joe Babstock wrote: There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. And how do you know it's a good book? I wouldn't mind a review and I may purchase the book (I doubt I qualify as a reviewer as I haven't yet figured this VoIP stuff out yet). I'm not really sure a few pages qualifies for a review. BTW, please excuse me if Paul is a frequent contributor to the mail list. I just found the method of announcement a bit suspect (I'm not say Paul posted this either). -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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: Some (lack of) answers regarding the wakeup call application...
Steve wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 06 July 2004 07:53 pm, Steve wrote: On Tuesday 06 July 2004 03:00 pm, Maron Kristófersson wrote: > Also, I need a Linux tool to splice a series of gsm audio > clips together in order to use one 'get_data' instead of multiple cat sound1.gsm > target.gsm cat sound2.gsm >> target.gsm Maron cat sound1.gsm sound2.gsm >>sound3.gsm is easier. Haha, should only have had one > A single > means create a new file (over writing the old one if possible) and a >> means append to the file (creating a new one if it doesn't exist). That a short description of it means, I'm sure I've missed a few details but that close enough for government work. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 Docs
Chris A. Icide wrote: On 08:10 AM 7/1/2004, Neil Cherry wrote: >> The WiKi is your friend. > >So far it hasn't been very friendly. I tried to to find a document >I printed out (it printed poorly). When I entered the document's >title it fail to list that link. I actually found it via google >(weird). I guess I need to learn a new way to think for searching >for Asterisk info. I'll learn. :-) > The Wiki's search system seems to leave a lot to be desired. I don't often use the search system in the wiki, I tend to either 'google' it, or if I know what I'm looking for, A few days ago I was looking for some info from the asterisk-addon cdr_odbc and I entered 'Asterisk cdr mysql". Well the search engine found the page I was looking for, and it was titled "Astersk cdr mysql", and yet, even though it was a perfect match, it was around #5 in the results. *shrug* references to pages of the voip-info.org wiki from google is your friend. I'm very good with google, I have it setup so I can type the search list into the url bar (or g and the list for things that look like a URL). There are also quite a few friends you can buy on asterisk-biz list, if you are so inclined. Nah, I'm actually trying to learn VoIP (yes the entire thing) and paying someone to do it won't help me learn. It's got to be learned by doing and search if you really want to know it. When I ask questions here I prefer pointers so I can learn to 'fish' so I can 'feed' myself. :-) -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 Docs
Kevin Walsh wrote:
Neil Cherry [EMAIL PROTECTED] wrote:
OK, this may seem to be an obvious question but where do I find
the reference docs? I'm getting this error message:
Timeout, but no rule 't' in context 'home'
about this line:
exten => 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know what the
parameters mean. I looking for a man page basically.
It has nothing to do with the 't' in your Dial().
The Dial() command docs can be found here:
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Ah, a key to the kingdom, thanks!
The "predefined extension names" list, including 't', can be found in
here:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
The 't' context is called when a timeout occurs. You could get rid
of the warning with the following:
exten => t,1,Hangup
That would simply hang up the line when a timeout is detected. You
could do anything you like in there, of course.
This page could be helpful too:
http://www.voip-info.org/wiki-Asterisk+cmd+ResponseTimeout
Thanks, it appears that I need to learn to use Wiki.
The WiKi is your friend.
So far it hasn't been very friendly. I tried to to find a document
I printed out (it printed poorly). When I entered the document's
title it fail to list that link. I actually found it via google
(weird). I guess I need to learn a new way to think for searching
for Asterisk info. I'll learn. :-)
My current set of problems are just configuration problems. I'm not
used to the commands, how they work and what they do. I accidently
figured out 't' after I got an error about no 'i' for invalid
extensions. Right now I'm wrestling with a SJPhone and the
Grandstream. Both have their own annoyances but I figure * will
be able to work around most of those.
BTW, let me say thanks. I don't want everyone to think I'm
just complaining. It's more frustration with the steep learning
curve.
--
Linux Home Automation Neil Cherry[EMAIL PROTECTED]
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[Asterisk-Users] Asterisk Docs
OK, this may seem to be an obvious question but where do I find
the reference docs? I'm getting this error message:
Timeout, but no rule 't' in context 'home'
about this line:
exten => 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know what the
parameters mean. I looking for a man page basically.
--
Linux Home Automation Neil Cherry[EMAIL PROTECTED]
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http://linuxha.sourceforge.net/ (SourceForge)
http://hcs.sourceforge.net/ (HCS II)
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Re: [Asterisk-Users] Do people actually answer questions here?
Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've only been watching this list for the past 2 days. And it seems to be an one way street: - -Tell about your problems and what you would like to do. Usually no answer. I have to admit I'm rather disappointed with Asterisk, information is probably available but very hard to find ; it seems to be limited to a few privileged people for whom their job is setting up VoIP system Uhm, that really sounds like VoIP anyway. I'm just starting (plenty of data but little if any VoIP). In order to get into this you have lots of due diligence to perform. The problem with VoIP is that it covers a lot of RFC's and topics (SIP, H323, MGCP, SKINNY, etc.) So far I haven't started asking questions because I haven't done enough home work yet. Leif gave some very good links. I also use google a lot. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 Cron
Samantha (Femtech) wrote:
Hi List
Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly
phonegc:/home/samantha# asterisk -r
Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to Asterisk CVS-05 currently running on phonegc (pid = 15938)
phonegc*CLI> restart now
phonegc*CLI>
Disconnected from Asterisk server
phonegc:/home/samantha#
This should work. It doesn't check to see if asterisk is running and
I haven't done any erro checking.
#!/usr/bin/expect
proc sleep {timeout} {
expect
}
set send_slow { 1 .05 } ;# How fast to exp_send the characters
#log_user 1 ;# Keeps the user from seeing the
;# spawned the echo back.
if [catch "spawn asterisk -r" reason ] {
exit 1
}
set id $spawn_id
expect "CLI>" { } ;# wait for a DOS prompt
exp_send -s -i $id "restart now\r" ;# Write out the table
sleep 2
exit 0
# End of code
###
--
Linux Home Automation Neil Cherry[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk with PostgreSQL
Caleb Kow wrote: Here we go: [EMAIL PROTECTED] root]# netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp0 0 *:32768 *:* LISTEN 3221/ tcp0 0 *:imaps *:* I didn't see Postgres running but did notice mysql. They run on different ports so that not a problem unless you are mistaking one for the other. Another poster stated that Postgres runs local socekts by default and that a change in the config is needed to get it working with TCP/IP. I'd investigate that as that's what it looks like. I hope this helps. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 with PostgreSQL
Caleb Kow wrote: Results of netstat -ap You seem to be missing the top part of the output which looks like this: [EMAIL PROTECTED] build]# netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp0 0 *:nfs *:* LISTEN - tcp0 0 *:time *:* LISTEN 3339/xinetd : : (more of the smae looking lines follow). Sorry if that wrapped. Active UNIX domain sockets (servers and established) Proto RefCnt Flags Type State I-Node PID/Program namePath unix 2 [ ACC ] STREAM LISTENING 5881 3623/ /tmp/.iroha_unix/IROHA unix 2 [ ACC ] STREAM LISTENING 3971 3326/ /var/lib/mysql/mysql.sock unix 2 [ ACC ] STREAM LISTENING 6002 3690/ /tmp/jd_sockV4 unix 2 [ ACC ] STREAM LISTENING 9522765 24900/httpd /var/run/fpcgisock unix 2 [ ACC ] STREAM LISTENING 5863 3607/gpm /dev/gpmctl unix 13 [ ] DGRAM2839 3171/syslogd /dev/log unix 2 [ ACC ] STREAM LISTENING 9623759 21382/postmaster/tmp/.s.PGSQL.5432 unix 2 [ ACC ] STREAM LISTENING 6060 3767/ /tmp/.font-unix/fs7100 unix 3 [ ] STREAM CONNECTED 9569452 3326/ /var/lib/mysql/mysql.sock unix 3 [ ] STREAM CONNECTED 9569451 24907/ unix 3 [ ] STREAM CONNECTED 9565499 3326/ /var/lib/mysql/mysql.sock unix 3 [ ] STREAM CONNECTED 9565498 24905/ unix 3 [ ] STREAM CONNECTED 9565496 3326/ /var/lib/mysql/mysql.sock unix 3 [ ] STREAM CONNECTED 9565495 24908/ unix 2 [ ] DGRAM9471559 9212/splogger unix 2 [ ] DGRAM5907 3634/crond unix 2 [ ] DGRAM5826 3598/spamd -d -c -a unix 2 [ ] DGRAM3642 3382/courierlogger unix 2 [ ] DGRAM3621 3371/courierlogger unix 2 [ ] DGRAM3600 3361/courierlogger unix 2 [ ] DGRAM3581 3350/courierlogger unix 2 [ ] DGRAM3225 3273/ unix 2 [ ] DGRAM3018 3262/xinetd unix 2 [ ] DGRAM2915 3221/ unix 2 [ ] DGRAM2847 3175/klogd On Thu, 24 Jun 2004 10:11:46 -0400, Neil Cherry <[EMAIL PROTECTED]> wrote: Caleb Kow wrote: Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql: got hostname of localhost Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:285 my_load_module: cdr_pgsql: got port of 5432 Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:288 my_load_module: cdr_pgsql: got user of asteriskpg Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:289 my_load_module: cdr_pgsql: got dbname of asteriskpgcdr Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:290 my_load_module: cdr_pgsql: got password of 65plesk Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? However, the strange thing is that when I try to connect to this database using the command prompt, it puts me through! :) Only when Asterisk tries to connect to the postgresql database does it not work. Any idea why this is happening? Can you do a netstat -ap ? -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 [EM
Re: [Asterisk-Users] Asterisk with PostgreSQL
Caleb Kow wrote: Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql: got hostname of localhost Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:285 my_load_module: cdr_pgsql: got port of 5432 Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:288 my_load_module: cdr_pgsql: got user of asteriskpg Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:289 my_load_module: cdr_pgsql: got dbname of asteriskpgcdr Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:290 my_load_module: cdr_pgsql: got password of 65plesk Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Jun 24 21:20:53 ERROR[1074494336]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? However, the strange thing is that when I try to connect to this database using the command prompt, it puts me through! :) Only when Asterisk tries to connect to the postgresql database does it not work. Any idea why this is happening? Can you do a netstat -ap ? -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 Noise
Brent Franks wrote: How does one prevent the interrupts from being shared? Check your BIOS settings. You should be able to assign from there. Do you mean like setting up the ISA slots? I've got a built in ethernet and USB which both sit on IRQ 10 (drives me nuts) and have no idea how to set either to another IRQ. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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 Noise
Ryan Courtnage wrote: On Wednesday 23 June 2004 08:17, Lee Norvall wrote: I have 2 x X100P on UK BT, both have been working fine for a while, but now I have started to get a beeping sound my end every 8/10 sec, and break-up in the voice call inbound/outbound. Any ideas??? Sounds like your x100p cards are sharing interrupts with another device. Check /proc/interrupts. How does one prevent the interrupts from being shared? -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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] Unable to find libiodbc.so.2
Manuel Wenger wrote: I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get that error: *CLI> load cdr_odbc.so Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Unable to load module cdr_odbc.so But the file is there... # ls -lag /usr/local/lib/libiodbc.so* lrwxrwxrwx1 root 17 Jun 22 15:23 /usr/local/lib/libiodbc.so -> libiodbc.so.2.1.9 lrwxrwxrwx1 root 17 Jun 22 15:23 /usr/local/lib/libiodbc.so.2 -> libiodbc.so.2.1.9 -rwxr-xr-x1 root 1448547 Jun 22 15:23 /usr/local/lib/libiodbc.so.2.1.9 Have you done an ldconfig (or reboot) since your last compile? -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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
