RE: [Asterisk-Users] Asterisk/Zaptel on Mac G5 or Xserve
Out of interest why use a G5 over an x86 PC? Do you feel the performance will be better, or do you just prefer Mac's? Thanks C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geoff Nordli Sent: 21 March 2005 20:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk/Zaptel on Mac G5 or Xserve I am considering a G5/XServe for a conferencing system. I need to put a TDM card in the machine for timing. Is anyone out there using Asterisk on Mac with zaptel drivers? I am looking at using Linux for the OS. If so what is your experience? Have a great day! Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] why even use SIP
And some kind of password to protect the config, so not any intelligent person with a prov. Utility can change your iaxy! Back on topic, I much prefer iax personally, however the lack of hardphone options is a real pain. Obviously depending on the situation, I use SIP hardphones on internal LAN, IAX softphone (firefly) on laptop for roaming, and all incoming and outgoing calls from * are sent via IAX. I use firefly with a Bluetooth headset for audio device and quality is stunning. However I agree firefly is not really upto it in a business environment. Regards C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson Sent: 22 March 2005 20:20 To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] why even use SIP Yeah, if we're making a list, add DNS name resolution to that list. :) -- Dana On Tue, 22 Mar 2005 23:48:46 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: If the IAXy had a bit more work done on it, it could be a good option, but it's not at the current time. Yep! Things like: - more codecs (just ulaw? come on...) - proper DHCP and possibility of static IP - a 'reset' button To start with would be nice to have. And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: VIA Mini-ITX, Asterisk, and hardware
Hi, I run * on the first 800mhz version they released. I do not use any PCI cards, so cannot coment on that I'm afraid. It works fine for testing in the environment I use...but I haven't stressed it at all. I had to make a change to the makefile for the processor, but I doubt that is needed for the newer versions. If I am right you made the cool little CF + flash disk * distro? I think they are an ideal pair. One of the new mini-itx boards comes with compact flash onboard, and has no builtin sockets except LAN and VGA. Very easy to make an embedded system. I think any of them 266 geode! C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: 20 March 2005 21:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OT: VIA Mini-ITX, Asterisk, and hardware Hello everyone, Does anyone out there have actual experience with running * on a mini-itx board from VIA? They look good, but I have some reserves because of VIA's problems with PCI latency in recent years (audio dropouts, wierd things happening). I am looking at the EPIA CL-1. For $270, I can get a CL-1 (1ghz C3, dual ethernets, etc), 256mb RAM, and a nice small (12 x 2 x 11) case (with 1 4cm fan)... They look like a good next step (or leap) up from a Soekris Net4801. I know that 1ghz C3 != 1ghz intel, but it's still probably better than a 266mhz Geode... I would love to try this board with Sangoma A101's, te110p's, and even some TDM4xx's, but if people out there already know that * is a bad fit here, I probably won't even bother and look elsewhere. Any tips, notes, caveats, etc from anyone? Anyone using any of the hardware I mentioned with one of these boards? Thanks! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Different codecs for different numbers via same IAX provider; how? Configs included.
Title: Different codecs for different numbers via same IAX provider; how? Configs included. Hi, I have been trying to work this out and havent been able to. I have some incoming numbers that come in over IAX, from the same server, and wish to use different codecs for different calls. This doesnt seem to work for incoming either. I cant seem to get any combination of allow/disallow to work. Ideally the following would work: [general] register = XX disallow=all [XXX] ;incoming number I want to use GSM with type= context= secret= allow=gsm [YY] ;incoming number I want to use alaw with type= context=XX secret=X allow=alaw However they only use one codec for both numbers. Am I doing something wrong in iax.conf, I am running stable, or is this something which * doesnt support yet. Thanks C ___ 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] Different codecs for different numbers via sameIAX provider; how? Configs included.
Sean, Thanks for the reply. I was afraid this was the problem. It looks like the called server can only negotiate on what codecs it wants to receive -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: 17 March 2005 20:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Different codecs for different numbers via sameIAX provider; how? Configs included. C. Tomlinson wrote: Hi, I have been trying to work this out and haven't been able to. I have some incoming numbers that come in over IAX, from the same server, and wish to use different codecs for different calls. This doesn't seem to work for incoming either. I cant seem to get any combination of allow/disallow to work.. Ideally the following would work: [general] register = XX disallow=all [XXX] ;incoming number I want to use GSM with type= context= secret= allow=gsm [YY] ;incoming number I want to use alaw with type= context=XX secret=X allow=alaw However they only use one codec for both numbers. Am I doing something wrong in iax.conf, I am running stable, or is this something which * doesn't support yet. Thanks C You will have to setup separate iax2 identities on the calling server ( ie: the server delivering the calls to you ) for this to work. Your receiving server has no way to negotiate a codec based on the incoming phone number ( correct me if I am wrong, but I see no way to do that ). Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced conference features, meetme2?
Hi, Thanks. I have already tried various options in from the wiki, but they don't work in my situation. I do not think the announce option works as I am using STABLE, not HEAD...huess I have to wait for it to make it into stable. I am creating dynamic conferences, using the 'd' option. The m option means everyone entering the conf can only listeni want selective mute(can get this via menu option, but not ideal) I will have a play with the admin option. Thanks C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 12 March 2005 23:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Advanced conference features, meetme2? Conference lock and member name been recorded and announced when they get in and out of a conference is already available. Check the wiki and look for meetme, you will see they have some parametes like m,a,s that will help you control this features. Anton Krall _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson Sent: Sábado, 12 de Marzo de 2005 12:31 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:[Asterisk-Users] Advanced conference features, meetme2? Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having the users name recorded before entering, and then played back to other callers on entrance etc etc. Are these things available now, or would they require development. Regards C File: ATT00137.txt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced conference features, meetme2?
Hi, Sorry, I relaise I could run head but didn't want to move across yet. I have found documentation on meetme fairly lacking; hence my n00bish questions. I realize you can get people to enter the same conferences but with different options; however I got stumped on a couple of things: Wiki informs me that agi scripts only work over Zap; I am using IAX or SIP' so I didn't look much further into them, I am unsure how meetme handles dtmf tones; my quick test didn't seem to work; namely sending a dtmf tone into a meetme conference DIDN'T send it to another extensions... I couldn't find, for example, a variable containing the current conference name. If I had those I agree it would be simple in the dialplan; just listen for a key eg 2, then when pressed kick user from conference, and immediately rejoin using a mute option, rejoining the same conference as the dialplan line got its info from the variable. Care to share any more infor e.g. on your AGI? Thanks C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: 13 March 2005 16:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Advanced conference features, meetme2? On Sun, 13 Mar 2005, C. Tomlinson wrote: Thanks. I have already tried various options in from the wiki, but they don't work in my situation. I do not think the announce option works as I am using STABLE, not HEAD...huess I have to wait for it to make it into stable. Or you can run cvs head. It is slated to become the next stable release not too far out. With a bit of testing it may be safe enough for your needs. Or not. :) I am creating dynamic conferences, using the 'd' option. The m option means everyone entering the conf can only listeni want selective mute(can get this via menu option, but not ideal) All the entrants do not have to use the same meetme command. We can have a mix of people entering muted/non-muted depending on their authorization. That's what the dialplan is there for. Actually, we do it through an agi now, but it used to be done in the dialplan itself. Now all the database stuff and all the prompts are handled through the agi. I think it is a mistake to add a lot of features to all the applications. Quite a few of these could have been implemented using the dialplan and a few new primitives. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced conference features, meetme2?
Peter, How does recording work..i file per person, or are they all muxed into one, or can you specify? What do you mean by new primitives? C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: 13 March 2005 21:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Advanced conference features, meetme2? On Sun, 13 Mar 2005, dean collins wrote: Taking yourself off mute is one of the more important requirements for broadcast conferences. That is available already: enable the star-menu with the 's' option. Entry 1 (the only one) allows the user to mute himself. I probably dial in to about 3 conference calls a week (using commercial services) where the default is everyone in the call is on mute and then you press star to talk - some automatically take you off or some flag it with the organizer to take you off mute. I'm prepared to pay a $25 bounty for this feature and automatically recorded conference calls feature. A speaker list is more advanced. Probably bet implemented with a manager interface to meetme that passes the dtmf digits to the manager interface and allows the manager interface to manipulate the conference. Recording is available in cvs head using the 'r' option. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on MS Virtual Server
I think you missed out a *NOT* below... In short, you *CANNOT* install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gilad Ben-Yossef Sent: 02 March 2005 14:59 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on MS Virtual Server Turgut Abacioglu wrote: Hello I downloaded Astwind and get working the network (means can access to Internet through MS Windows). DEbian and Asterisk files are updated from Internet. But When I make install in Zaptel (it was my first make) I got many errors. Acoording to one manual this happens when we do not have modeversion .h kernel header file (according to it, it should reside in /usr/src/linux) which in /usr/src/linux, a make menuconfig will create it. BuT I do not have the linux dir (in /usr/src) and kernel source files thus modversion.h file. In addition I do not know how to download kernel files to linux directory (I tried apt-get but I could not format properly the /etc/spt/source.list file) Could you help. Am I in the correct path? No, you are not. Zaptel is a driver to hardware cards. CoLinux (on which Astwind is based) is a virtual Linux running as a Windows task. Virtual here means - no hardware. In short, you can install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Advanced conference features, meetme2?
Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having the users name recorded before entering, and then played back to other callers on entrance etc etc. Are these things available now, or would they require development. Regards C attachment: winmail.dat___ 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] AAH 0.06 - IAX Connection Over NAT Firewall
Title: [Asterisk-Users] SIP to H.323 no audio As I understand it if you use that deny statement, all calls will be disallowed, hence why you couldnt get any incoming calls. If you add an allow line with the VOIP providers IP that it send the call from, you can then use that line to disallow everything else. It is just a security feature really. C From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: 11 March 2005 07:23 To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Digium. Com Subject: RE: [Asterisk-Users] AAH 0.06 - IAX Connection Over NAT Firewall OK. I removed the deny statement they have me using and now I can get incoming calls. Do I need the deny 0.0.0.0/0.0.0.0 statement? Thanks, Wiley From: [EMAIL PROTECTED] on behalf of Wiley Siler Sent: Thu 3/10/2005 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] AAH 0.06 - IAX Connection Over NAT Firewall Hello all, I am having trouble getting my IAX based Voip provider setup. Any pointers are welcome. So here is the deal. I am registered up and I can make outgoing calls but incoming calls fail. Configs all look good I thought. My PBX is behind our firewall with a direct NAT of one to one for an external IP. IAX port is forwarded UDP and TCP to the internal IP. * shows good registration and Ips and ports show solid. Within my AAH I have the registration like the provier said to do. I get absolutely nothing on the incoming. IAX2 debug shows nothing on incoming. Just a fast busy. Outgoing works perfectly however. I have a defined DID in the AMP interface and verified it is written to confs and have reloaded. Can anyone tell me another way to verify that something is coming in? Or did I just miss something on the whole IAX over NAT? Thanks all, Wiley ___ 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] Problem with incoming calls.
Title: Problem with incoming calls. I have a problem with incoming IAX calls. I have 2 numbers from the same supplier delivered over IAX. I register once with the server, and both calls get to my box, and I get output on the console with both calls. However I cannot get each number to go to separate contexts Please see relevant sections from extensions and iax conf. files Section from IAX.conf: [448700XX] ;incoming 0870 number type=user username=448700XX context=conference trunking=off [448450XX] ;incoming 0845 number type=user username=448450XX context=demo_default trunking=off Section from extensions.conf [demo_default] ;the 0845 number should go here exten = 448450XX,1,Answer exten = ..i have more here. [conference] ;the 0870 number should go here exten = 448700XX,1,Answer exten = ..i have more here. The output on the CLI looks like: NOTICE[1282]: chan_iax2.c:5461 socket_read: Rejected connect attempt from XXX.XXX.X.XXX, request '[EMAIL PROTECTED]' does not exist However if you look above, the 0870 number should go to the [conference] context; not the [demo_default] one.. If I then call the other 0845 number it works: -- Accepting unauthenticated call from XXX.XXX.X.XXX, requested format = 8, actual format = 8 -working If I comment out one of the numbers in the iax.conf, the other one works fine.its just when both are active it doesnt seem to play properly. Does anyone have any ideas? As far as I know I'm not being stupid, but please point it out if I am. Any help much appreciated. Regards, C ___ 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 incoming calls.
Title: Problem with incoming calls. I now have a workaround for this problem; I carried on research after posting. I now route both incoming numbers in the IAX.conf into one context, like they wanted to before. In the new context I just use the goto command to farm the numbers out to separate contexts. This works well. I believe this problem is due to both number being from the same account; which it only want to associate with one context. Regards C From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson Sent: 10 March 2005 13:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Problem with incoming calls. I have a problem with incoming IAX calls. I have 2 numbers from the same supplier delivered over IAX. I register once with the server, and both calls get to my box, and I get output on the console with both calls. However I cannot get each number to go to separate contexts Please see relevant sections from extensions and iax conf. files Section from IAX.conf: [448700XX] ;incoming 0870 number type=user username=448700XX context=conference trunking=off [448450XX] ;incoming 0845 number type=user username=448450XX context=demo_default trunking=off Section from extensions.conf [demo_default] ;the 0845 number should go here exten = 448450XX,1,Answer exten = ..i have more here. [conference] ;the 0870 number should go here exten = 448700XX,1,Answer exten = ..i have more here. The output on the CLI looks like: NOTICE[1282]: chan_iax2.c:5461 socket_read: Rejected connect attempt from XXX.XXX.X.XXX, request '[EMAIL PROTECTED]' does not exist However if you look above, the 0870 number should go to the [conference] context; not the [demo_default] one.. If I then call the other 0845 number it works: -- Accepting unauthenticated call from XXX.XXX.X.XXX, requested format = 8, actual format = 8 -working If I comment out one of the numbers in the iax.conf, the other one works fine.its just when both are active it doesnt seem to play properly. Does anyone have any ideas? As far as I know I'm not being stupid, but please point it out if I am. Any help much appreciated. Regards, C ___ 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: Do I Need Astrisk
Hi, Do you mind me asking for a little more information regarding what this would be used for? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rafal Sent: 10 March 2005 18:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Do I Need Astrisk Hi, can someone knowledgebale about pbx's please help mewith some research before I approach people for help/funding.. (its for a youth/community web site in the uk) I require a pbx that: 1) on dialing in: -requests a user ID, checks adatabase if theuserID callerID matches -records a message transcoding it to a streaming audio format -logs the call;user ID; address of the media file in database 2) dials out: -from database -plays streaming file down phone or onto voicemail what are my options here? please advise.. (my knowledge level is basic python general scripting) tia -- Rafal Kaniewski EU:UK ___ 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] OT: AstLinux 0.2.2 released
This looks a great little distro, I will definitely try it if I have the need for a solid state machine. Most modern systems are capable of booting from USB stick; would this be possible, instead of CF? One stick for astlinux, and another for conf files? It strikes me that the mini-itx via systems might be a good alternative if the soekris board is not available to people? Regards, C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: 10 March 2005 22:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: AstLinux 0.2.2 released Jeb Campbell wrote: Kristian Kielhofner wrote: Hello Everyone, I have released AstLinux 0.2.2. There are way too many improvements to list here, but here is a short summary: Linux 2.4.27, iptables, mini_httpd (with PHP SSL), phpconfig, AstShape traffic shaping, tftp server, OpenSSH, proftpd, Soekris Net4801 and Pentium-MMX and higher x86 support. There is actually WAY more software, but I couldn't possibly list it all. It is now available as a Windows install package (or 32mb Compact Flash image). AstLinux 0.2.2 occupies around 26mb of disk space once expanded to flash. The gzip'd CF images are about 15mb and the Windows installer isn't much more (for both images, the PDF user guide, and a copy of Putty). Wow -- very impressive. I have it downloaded but can't install it yet (at work now). I was working on the same thing and using a Gentoo build system for a 2.6 kernel and uclibc, but then went to glibc -- and now have a full system (perl, python, etc) in ~57M. Anyway I was just wandering if you had your build sources/scripts online so that people could customize Astlinux? I for one would like to be using the stable cvs. Thanks, Jeb Campbell [EMAIL PROTECTED] Jeb, The source tree is pretty rambunctious. It exists in a directory on one of my systems and things are kind of all over the place. Now that I have finally released 0.2, I can work at cleaning it up. I actually have never used CVS, I guess I have just never really felt that I needed to... Maybe I'm just ignorant, but there really isn't a whole lot of source to this project, so CVS seems like overkill. It was made with PTXDist, Crosstool, and a few other OSS packages not available from PTXDist (like Asterisk). The source could probably be provided in the form of Crosstool scripts, PTXDist config files, and maybe some patches to the other packages like Asterisk. I will decide on this and provide that stuff soon. The real work was getting it all to run from Compact Flash in a sane manner - and my shell scripts in AstLinux are what accomplish that feat. My old builds included Perl and Python, but I just didn't see the need for them in such a targeted distro (if you even want to call it that). They seem more suited for general-purpose distros. But 57mb is not bad for all of that... If you are running AstLinux 0.2.2 and use the astup update script, you will basically be running stable CVS'. That script will rsync your system against my public rsync server to bring you up to the latest feature set. Make sure to read the User Guide to understand how all of these pieces work together. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJphone on PDA registering with Asterisk???
Ronald, You will need to give *more* information than that I have SJphone on my PDA, and have setup a SIP account on *, and it works fine :-) I take it you have setup sjphone to register to *. I take it your PDA has a network connection? C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: 06 March 2005 14:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SJphone on PDA registering with Asterisk??? I try to setup SJphone on my PDA, but it does not register with Asterisk. I have setup a sip account on asterisk, ... Can anybody give me a hint? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Hi Again, I used my phpconfig setup for a week, and found it a great timesaver for me :-) However I have just gone and broken it, and can't seem to 'fix' it. I was running a xorcom rapid installation, but converted to a semi-standard debian by changing the apt sources; so I could install a couple of extra things. I did apt-setup and choose British FTP sites I did apt-get dist-upgrade which installed a lot of stuff However this changed my box rather more than I was expecting for example it installed caudium as a web server, when I already have apache. I removed that via apt-get --purge remove caudium And after a couple of tweaks, I now have apache running again fine (as far as I can tell), and all my other web things work fine. However now I cannot even browse a .conf file via phpconfig. When clicking on the file I get the following error: Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission denied in /var/www/phpconfig/cls_phpconfig.php on line 127 I have gone over the wiki page, done chmod again etc, but nothing makes a difference. Does anybody have any ideas? Thanks C ___ 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] FW: Getting PHP Config to work?
Thanks, I had thought this, and done the command: chmod -R a+w /etc/asterisk And it still didn't work. However I just set chmod 777 via WinSCP recursively, and it worked :) This is only a testing box I am not worried about the security risks. Strange the chmod didn't work I feel? C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: 05 March 2005 12:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? However now I cannot even browse a .conf file via phpconfig. When clicking on the file I get the following error: Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission denied in /var/www/phpconfig/cls_phpconfig.php on line 127 I have gone over the wiki page, done chmod again etc, but nothing makes a difference. Your apache doesn't have read access on the file. It can't read the file or even worse, it can't go in that dir. Check that /etc/asterisk is readable (and writable) by apache. Also check that the conf files are readable by apache. hth ___ 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] Getting asterisk-addons installed on Debian?
Title: Getting asterisk-addons installed on Debian? Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. I have mysql and mysql-devel packages installed I think!? pbx01:/usr/src/asterisk-addons# dpkg -l mysql-server libmysqlclient*dev Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed |/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err: uppercase=bad) ||/ Name Version Description +++-===-===-== ii mysql-server 4.0.23-7 mysql database server binaries un libmysqlclient-dev none (no description available) pn libmysqlclient10-de none (no description available) ii libmysqlclient12-de 4.0.23-7 mysql database development files un libmysqlclient14-de none (no description available) un libmysqlclient6-dev none (no description available) un libmysqlclient9-dev none (no description available) pbx01:/usr/src/asterisk-addons# Which I know you need. I have mysql running etc. The problem seems to be making asterisk-addons I have exported from the cvs and tried both CVS and STABLE versions. I am running asterisk stable, installed via xorcom rapid (which may be why it freaks out?) When googling I didnt find much, bar one similar problem with no replies. Output: pbx01:/usr/src/asterisk-addons# make clean rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' pbx01:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` ./mkdep: line 85: cc: command not found make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o common.o common.c make[1]: gcc: Command not found make[1]: *** [common.o] Error 127 make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 pbx01:/usr/src/asterisk-addons# A few errors.. If anyone could help with any easy way to install asterisk-addons, or just the mysql section, that would be great. I havent been able to find ___ 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] Getting asterisk-addons installed on Debian?
Hi, Thanks. I idd that and now get different errors: pbx01:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:17:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:18:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:19:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:20:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:21:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:22:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 pbx01:/usr/src/asterisk-addons# Now this is probably due to me not having compiled * to start with, so I have no /usr/src/asterisk folder. I feel I may be better starting from scratch with a default Debian installation, and then I will know what I have where? What are your opinions? The best thing would be an apt-get install asterisk-addons, but I haven't found that :/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martijn van Oosterhout Sent: 05 March 2005 13:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Getting asterisk-addons installed on Debian? On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote: Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. snip make[1]: gcc: Command not found You need a C compiler, try apt-get install build-essential Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk for Live-Stream?
Something like this sis similar to what you are looking for I think. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Felix E. Klee Sent: 05 March 2005 17:17 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk for Live-Stream? I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event -- mobile phone -- software answering machine -- Internet server * Event -- mobile phone -- VOIP -- Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would Asterisk fit the bill? Alternatives? [1] It's Monday's EU Council of Ministers with Software Patents on the agenda: http://wiki.ffii.org/Dkparl050304En -- Felix E. Klee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing from a website. How to start...?
Alistair, I may be confused, but I thought this was a users list, not really for advertising business activity? The last 2 posts of yours have been blatant adverts for your business..could you not take them off list? I may be fairly new here and may havegot the wrong impression, if so I'm sure I will be corrected. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: 04 March 2005 11:19 To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialing from a website. How to start...? Evert, The best way to do this is have your PHP code put a control file in the outgoing directory of Asterisk. This then invokes an Asterisk macro that calls the user, then transfers them to the contact. The format of the file is at: http://www.voip-info.org/wiki-Asterisk+auto-dial+out I run a consulting firm doing (amongst other things) Asterisk work. If you're interested, we can install Asterisk, configure it to talk to your telephone system, set up the click to dial, and integrate it with your PHP - email me off list. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Evert Meulie wrote: Hi all! We use a PHP-portal for management of our projects contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so I was wondering whether there's someone who can point me in the right direction... :-) (Preferable in PHP, since that's the flavor of choice of our portal) Regards, Evert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What my IAXy could have been...
Sounds awesome. Hadn't come across these before, lots of interesting possibilities. Do you have a link to the IAX project? I found http://sourceforge.net/mailarchive/forum.php?thread_id=6720059forum_id=3894 0 which was the most informative. Only a couple of mention on this list. I take it updates to the Iaxy have stopped? C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nik Martin Sent: 03 March 2005 17:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] What my IAXy could have been... Matt Schulte wrote: you and everyone else :-) From: Daiku [mailto:[EMAIL PROTECTED] But i AM looking for info on another IAX capable device - like the IAXy, but more user friendly, as it were... http://www.gumstix.com There's a grass roots IAX based phone starting up using these awesome Linux boxes. BOA web server, IAXcomm, speech recognition, bluetooth headset, etc. Really nice, and a chance to build the IAXy you always wanted. Nik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What my IAXy could have been...
Hi, What are your opinions on the iaxy? I have one coming. From what I have seen, at least in the uk, iax2 hardphones are NOT widespread; iaxtalk.com are the only store I can find which sell them? C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daiku Sent: 01 March 2005 13:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What my IAXy could have been... Hi, methinks that in the good 3 months since i ordered an IAXy, things have changed so much that now almost anybody out there with a VoIP hardweare website offers complete phones for less money than the IAXy, with support for both IAX2 and SIP in many cases, and fully configurable via its own keypad. What would you recommend as a rugged, small, and easy to configure/use self-contained unit that one could carry along in one's hand luggage when traveling and plug into the LAN wherever someone has an ADSL, cable, or fibre-optic connection? A VoIP unit for all seasons? Thanks regards: Hendrik -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mini atx and asterisk (EPIA and the like)
If you are talking about the epia etc boards, they are mini ITX.. I am running an 800mhz one with 256mb ram as a test server, purely voip, using a couple of SIP and IAX clients. No moh yet. I had to modify the makefile in order for it to work, but once working its fine so far. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: 01 March 2005 14:42 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] mini atx and asterisk (EPIA and the like) Hi, haven't found anything in google's, i wonder if there is a comparative page of what to expect from running * on motherboards like the EPIA and similar ones. Since i have not used *ever* such kind of mini atx form factor boards, I have no clue about their performance. SIP-SIP communications, voicemail SIP-TDM communications, voicemail how may users (SIP hardphones and analog phones via CPE equipment) Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Hi, Its now up at http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig I would be interested in any feedback. Hope it helps. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 28 February 2005 04:50 To: C. Tomlinson Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I'll look out for it, thanks! Julius. Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 25 February 2005 14:33 To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I am having trouble using cvs, is it possible to use cvsup or any other method available and still get to install, configure and use phpconfig? If so, how do I go about it? Julius. Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I'll look out for it, thanks! Julius. Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO ___ 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] phpconfig
Hi, The default install from that turorial gave me fully functioning links etc. What format are your config files in; care to post an extract? What version of PHP are you running? Regards, C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allan hank Sent: 28 February 2005 15:23 To: Time Bandit Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] phpconfig Hello, That's the document i read and got all the relevant links. I also tried to follow all the predures . More help is appreciated, Thanks very much Allan On Mon, 28 Feb 2005 10:13:02 -0500, Time Bandit [EMAIL PROTECTED] wrote: Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? See this newly created document, it explains everything you need to make it work. http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig It's been written with the help of peoples on this list. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Behind NAT
I know this is possible using IAX easily, although I guess that is not an option for you. I have no firsthand experience, but believe some have got it working via careful setup e.g noreinvites and other things. If you setup a linux router, you could maybe have a separate DMZ to the * box, but still use QoS? Hope someone else can help you. Best of luck with the new job. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sammy ominsky Sent: 28 February 2005 21:04 To: Asterisk Users Subject: [Asterisk-Users] Asterisk Behind NAT Hi all, I've done quite a bit of reading, and I see that it's going to be difficult, but as a last-ditch effort before implementing a suggestion I don't like at all, I figured I'd ask... Has anyone successfully put an asterisk box on an internal network behind a NAT device and been able to connect with SIP from outside? The real point behind all this is to implement QoS for the voice traffic, and putting a third box in front of the asterisk and NAT boxes has been deemed too expensive. Currently, asterisk has a public IP, as does the NAT box behind which all the office machines sit. If it can be done, the NAT box would be the best place to do the QoS, so why not ask, right? Alternatively, I'm open to any suggestions that would work. I've been handed this challenge on my first day on a new job... :/ Thanks, ---sambo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home
Title: [EMAIL PROTECTED] There is a forum type thing on its sourceforge page http://sourceforge.net/projects/asteriskathome/ under the forum tab. Hope it helps. C From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Seddon Sent: 27 February 2005 14:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] Is there a forum for [EMAIL PROTECTED] Its a great Asterisk option but I have some questions and this forum doesnt seem the right place to ask. Bill Seddon Lyquidity Solutions Limited ___ 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] DISA and a long delay; ideas?
Title: DISA and a long delay; ideas? Hi, I have just setup a DISA setup whereby people can dial in, authenticate, are given a dialtone and can then call out. Everything works however there is a 10 second delay after the user enters the number and presses #, until the system does anything. Here is the relevant section from my extensions.conf: ___ 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] DISA and a long delay; ideas?
Title: DISA and a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate,
are given a dialtone and can then call out.
Everything works however there
is a 10 second delay after the user enters the number and presses #, until the
system does anything.
Here is the relevant section from my extensions.conf:
[dialtone]
exten = s,1,Authenticate(1234)
exten = s,2,DISA(no-password|dialtone_outgoing)
[dialtone_outgoing]
exten = _01.,1,Dial(${OUTGOING}/44${EXTEN:1},30,L(6:3:1))
exten = _07.,1,Playback(pbx-invalid)
The call gets dropped into the dialtone
context fine.
The authentication works fine
The user is given a dialtone fine.
If the user dials a 01 number, the call
goes out.
If the user dials an 07 number, then the
message plays back.
HOWEVER there is a 10 second delay between
the dialing (followed by #) and the system doing anything.
Watching the log with verbosity set to 16,
no messages occur in the 10seconds.
Any ideas; do I have the dialplan correct?
Thanks
C
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson
Sent: 27 February 2005 17:10
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] DISA and
a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate,
are given a dialtone and can then call out.
Everything works however there
is a 10 second delay after the user enters the number and presses #, until the
system does anything.
Here is the relevant section from my extensions.conf:
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RE: [Asterisk-Users] FW: Getting PHP Config to work?
Hi, I just tried your config setup out, seemed to work great. I guess the reload scripts etc are a work in progress :p I could edit files just fine though, threw the scrips in my /www dir and didn't tweak anything else, I guess its all done due to my phpconfig installation. Drop us a line if you do update them, although between phpconfig and this I should be fine :-) C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: 25 February 2005 14:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. I had the same problem. So I did a simple PHP page that let me do this. You can grab it here : http://www.marccharbonneau.com/asterisk/asweadto_0_1_1.tar That was before I discovered phpconfig. That doesn't say I won't continue working on mine :) However I cannot write any files, I get the error: User: admindoes not have access to this feature. Write failed! I found some errors in phpconfig. Open the file cls_phpconfig.php In the function OC_readConfFile around line 131 change : $this-_OC_the_file[] = fgetc($file); to : $this-_OC_the_file[] = fgets($file); In the function OC_checkAccess around line 438 change : $accessFile[] = fgetc($file); to : $accessFile[] = fgets($file); fgetc read one character at a time. fgets read one line at a time. I have moved asterisk.reload into /bin, and if I run it from the shell I get You don't have to move it to /bin. You can just do this simple modification to have it run from the same place as the pages Open the file phpconfig.php Look for : $reset_cmd = asterisk.reload and change to $reset_cmd = ./asterisk.reload You should be running fine with this. If not, let me know, I may have forgot something ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DISA and a long delay; ideas?
Many thanks, that was the problem.
I didn't paste the context that forwards the call into the DISA context; it
had this in:
...DigitTimeout,5
..ResponseTimeout,10
Doh!
It works great with the mobile number, as I can pattern match 10 digits:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: 27 February 2005 21:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DISA and a long delay; ideas?
On Sun, 27 Feb 2005, C. Tomlinson wrote:
I have just setup a DISA setup whereby people can dial in, authenticate,
are
given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters
the number and presses #, until the system does anything.
Here is the relevant section from my extensions.conf:
[dialtone]
exten = s,1,Authenticate(1234)
exten = s,2,DISA(no-password|dialtone_outgoing)
[dialtone_outgoing]
exten = _01.,1,Dial(${OUTGOING}/44${EXTEN:1},30,L(6:3:1))
exten = _07.,1,Playback(pbx-invalid)
snip
HOWEVER there is a 10 second delay between the dialing (followed by #) and
the system doing anything.
My first guess would be digit timeouts. Your patterns are _01. and _07..
These don't give asterisk any hints about how many digits to expect, so
its only choice is to wait for the maximum digit timeout period to be sure
that it doesn't make a decision early before you've entered all your
digits.
The best thing (in my view) would be to completely specify the digit
patterns you want users to be able to use. This gives you the opportunity
to control which numbers may be called and which may not, and it also
gives asterisk hints about what kinds of digit patterns it should expect.
These hints allow it to make faster decisions about whether a digit
pattern is complete and/or valid. An alternative would be to use the
DigitTimeout application to set a lower timeout period.
Greg
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FW: [Asterisk-Users] DISA and a long delay; ideas?
Jeez, I need to work out the shortcut to send an email which I keep pressing
by accident!!
-Original Message-
From: C. Tomlinson [mailto:[EMAIL PROTECTED]
Sent: 27 February 2005 22:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DISA and a long delay; ideas?
Many thanks, that was the problem.
I didn't paste the context that forwards the call into the DISA context; it
had this in:
...DigitTimeout,5
..ResponseTimeout,10
Doh!
It works great with the mobile number, as I can pattern match 10 digits:
exten = _07X,1,Playback(pbx-invalid)
However I need to play with the 01 and 02 numbers as they can either be 10
or 11 digits I think (Uk landline numbers can be either?!)
Think I need to read this
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
And work out the best way to proceed...
If I match 10 digits it immediately dials, even if I want 11. I don't really
want a delay, even if I set timeout down to 5 or something!
Cheers
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: 27 February 2005 21:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DISA and a long delay; ideas?
On Sun, 27 Feb 2005, C. Tomlinson wrote:
I have just setup a DISA setup whereby people can dial in, authenticate,
are
given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters
the number and presses #, until the system does anything.
Here is the relevant section from my extensions.conf:
[dialtone]
exten = s,1,Authenticate(1234)
exten = s,2,DISA(no-password|dialtone_outgoing)
[dialtone_outgoing]
exten = _01.,1,Dial(${OUTGOING}/44${EXTEN:1},30,L(6:3:1))
exten = _07.,1,Playback(pbx-invalid)
snip
HOWEVER there is a 10 second delay between the dialing (followed by #) and
the system doing anything.
My first guess would be digit timeouts. Your patterns are _01. and _07..
These don't give asterisk any hints about how many digits to expect, so
its only choice is to wait for the maximum digit timeout period to be sure
that it doesn't make a decision early before you've entered all your
digits.
The best thing (in my view) would be to completely specify the digit
patterns you want users to be able to use. This gives you the opportunity
to control which numbers may be called and which may not, and it also
gives asterisk hints about what kinds of digit patterns it should expect.
These hints allow it to make faster decisions about whether a digit
pattern is complete and/or valid. An alternative would be to use the
DigitTimeout application to set a lower timeout period.
Greg
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RE: [Asterisk-Users] DISA and a long delay; ideas?
Hi, I am running Asterisk 1.0.5 Stable, and changing the pattern matching e.g to 10 digits made it call out instantly :-) Not sure what your problem is. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: 27 February 2005 22:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DISA and a long delay; ideas? On Mon, February 28, 2005 8:33, Rod Bacon said: I agree. The following commands may also be of use... Actually I disagree, I'm running 2 different asterisk servers, one with 1.0.5 and the other with CVS and I noticed this last night, the cvs version attempts to send within a reasonable time, but the stable version waits a long time before attempting to send, the strange thing is the dial plans for disa are identical and it was the same sipura connecting to both... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
I had a look but $100 seems a bit steep to me at the minute! C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herman Cremer Sent: 25 February 2005 14:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? If you are using windows, have a look at Zend Studio that is used for PHP but can do wonders for other editing apps as well. -herman On Fri, 2005-02-25 at 15:52, C. Tomlinson wrote: Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. Thanks Again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Folwell Sent: 25 February 2005 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Hi Tzafrir, I do accept that there are many security issues with this setup. However I agree with the post in the previous thread: If the asterisk server is reachable from the outside over http or other unsecure protocols, it would be really dangerous. But in a trusty intranet-environment, where firewalls block every attempt to access the asterisk server from the outside, this solution should be save enough, even if nothing is really save enough ;-) . -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 25 February 2005 18:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote: Hi, Thanks for the batchfile type, it's a handy one. I'm not editing over the internet, just local LAN for testing ATM. Protected via firewall. Would it not be fairly secure using a combination of the following: .htaccess file VPN? https access? Limit apache to only allow certain IP's? And the public keys thing. Secure agains what? What are the threats you consider? VPN and/or limit of IP addresses (in iptables or in apache's config) would serve to allow access only from certain addresses. But is this a relaistic limitation? I thout you wanted to be able to edit the configuration from various hosts. If this is only your setup, then an sftp-based setup is probably more convinient. Using a .htaccess file (or even better: an apache config snippet in /etc/apache/conf.d )you can force authentication to get to this directory. But then-again, you empower the apache user (www-data) to configure and control asterisk, and thus if anybody manages to make your web server execute an arbitrary script (e.g: can write to a .php file under the wwwroot) they can make asterisk execute arbitrary code as well. The chmod command makes Asterisk's configuration world-writable. So anybody with temporary write access to your system can again change asterisk's configuration. This breaks a general sanity assumption that only system users can write to the configuration. As a rule of thumb such a chmod should generally be replaced by adding a certain user to a certain group. You also change the permissions to the asterisk reload script to 777. Why does it need to be world-writable? This gives an attacker yet another place to inject executable code. In short: I still fail to see the atvantages of using phpconfig in your settings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] FW: Getting PHP Config to work?
-Original Message- From: C. Tomlinson [mailto:[EMAIL PROTECTED] Sent: 26 February 2005 11:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? Hi Tzafrir, I do accept that there are many security issues with this setup. I am fairly ignorant of the exact problems due to my lack of knowledge. However I agree with the post in the previous thread: If the asterisk server is reachable from the outside over http or other unsecure protocols, it would be really dangerous. But in a trusty intranet-environment, where firewalls block every attempt to access the asterisk server from the outside, this solution should be save enough, even if nothing is really save enough ;-) . Guido Hecken What exactly do you mean by an sftp based setup? Is this like the builtin editor in WinSCP? Phpconfig allows me to change the config by any pc on my LAN, using windows, mac, pocket pc(have to test this one) etc. This is handy for me for testing. I like the flexibility it gives me. The * box is behind a NAT firewall, the only ports open being those for IAX. If I setup a VPN in the future I will be able to access the phpconfig files securely (?) via that. It may not suite everyone. Maybe the 777 CHMOD could be done better, but this was the way it worked for me. I am fairly new to Linux and *, so my methods will not be the best. Thanks for all the informationif I get to a production box I will probably not use phpconfig! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 25 February 2005 18:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote: Hi, Thanks for the batchfile type, it's a handy one. I'm not editing over the internet, just local LAN for testing ATM. Protected via firewall. Would it not be fairly secure using a combination of the following: .htaccess file VPN? https access? Limit apache to only allow certain IP's? And the public keys thing. Secure agains what? What are the threats you consider? VPN and/or limit of IP addresses (in iptables or in apache's config) would serve to allow access only from certain addresses. But is this a relaistic limitation? I thout you wanted to be able to edit the configuration from various hosts. If this is only your setup, then an sftp-based setup is probably more convinient. Using a .htaccess file (or even better: an apache config snippet in /etc/apache/conf.d )you can force authentication to get to this directory. But then-again, you empower the apache user (www-data) to configure and control asterisk, and thus if anybody manages to make your web server execute an arbitrary script (e.g: can write to a .php file under the wwwroot) they can make asterisk execute arbitrary code as well. The chmod command makes Asterisk's configuration world-writable. So anybody with temporary write access to your system can again change asterisk's configuration. This breaks a general sanity assumption that only system users can write to the configuration. As a rule of thumb such a chmod should generally be replaced by adding a certain user to a certain group. You also change the permissions to the asterisk reload script to 777. Why does it need to be world-writable? This gives an attacker yet another place to inject executable code. In short: I still fail to see the atvantages of using phpconfig in your settings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] FW: Getting PHP Config to work?
Hi, I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. Hence phpconfig would be great, however I am having difficulty getting it to work. I have searched the message boards and the wiki, and found nothing of help for this problem :( I have a full working apache/php setup (default install) and have added the phpconfig files to the www dir, and they are accessible over the LAN. So far so good. I Can read the files fine. However I cannot write any files, I get the error: User: admindoes not have access to this feature. Write failed! I tried messing with the CHMOD settings of the files but no joy. My manager.conf looks like: ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = secret ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.73/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user I can successfully telnet into the manager interface through shell on the local machine and winxp machine on the LAN I have moved asterisk.reload into /bin, and if I run it from the shell I get a successful? Output: pbx01:~# /bin/asterisk.reload Asterisk Call Manager/1.0 Can anyone help? It is the same error the online example gives. Is it something to do with specific admin rights in xorcom, or have I missed something fundamentally wrong out? I have checked the php files and the paths seem to be OK (default * installs) I have a couple of ideas as to the problem: -PHP needs something enabled e.g safemode? -Xorcom has changed something phpconfig needs e.g * not running as root or something? Many Thanks, C ___ 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
Mark, Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: 25 February 2005 13:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem On Wed, 2005-02-23 at 14:22 +0100, Roberto Piola wrote: We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. Thanks Again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Folwell Sent: 25 February 2005 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
I am going to try out all the instructions and document it, and then submit to the wiki so future installations are easier for all :-) I will post the draft 1st here. Thanks for the help, lets hope I get it working. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hecken, Guido Sent: 25 February 2005 15:15 To: Time Bandit; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? I found some errors in phpconfig. Open the file cls_phpconfig.php In the function OC_readConfFile around line 131 change : $this-_OC_the_file[] = fgetc($file); to : $this-_OC_the_file[] = fgets($file); In the function OC_checkAccess around line 438 change : $accessFile[] = fgetc($file); to : $accessFile[] = fgets($file); fgetc read one character at a time. fgets read one line at a time. I have moved asterisk.reload into /bin, and if I run it from the shell I get You don't have to move it to /bin. You can just do this simple modification to have it run from the same place as the pages Open the file phpconfig.php Look for : $reset_cmd = asterisk.reload and change to $reset_cmd = ./asterisk.reload Some time ago, I had the same probs with phpconfig and had to search and google quite a long time to get it running. Since our systems are now running fine with phpconfig, I simply forgot the above fgetc/fgets issue. Therefore... A wonderful place for all this would be the wiki ;-) Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 25 February 2005 14:33 To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I am having trouble using cvs, is it possible to use cvsup or any other method available and still get to install, configure and use phpconfig? If so, how do I go about it? Julius. Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Hi, I'm not sure the way to change it, but when I d/l it from http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig/cls_phpconfig.ph p?login=2asterisksess=5c8e63576772790cfc2e1dbce354e04d I had read about the problem with fget's, but presumed this change was the correct one. However it looks like my skim reading got the better of me! I am writing up an installation guide now. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: 25 February 2005 15:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? Some time ago, I had the same probs with phpconfig and had to search and google quite a long time to get it running. Since our systems are now running fine with phpconfig, I simply forgot the above fgetc/fgets issue. Therefore... A wonderful place for all this would be the wiki ;-) Better yet, update the CVS with the correction. How would I go about that ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Thanks to all your help I have now got it working great. I have written a quick howto which I plan to add to the wiki if people approve? Take a look at: http://www.burntwires.com/asterisk/Install%20PHP%20Config.htm (Please excuse the bloated html) Please leave any feedback and then I will add to the wiki. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson Sent: 25 February 2005 13:32 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FW: Getting PHP Config to work? Hi, I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. Hence phpconfig would be great, however I am having difficulty getting it to work. I have searched the message boards and the wiki, and found nothing of help for this problem :( I have a full working apache/php setup (default install) and have added the phpconfig files to the www dir, and they are accessible over the LAN. So far so good. I Can read the files fine. However I cannot write any files, I get the error: User: admindoes not have access to this feature. Write failed! I tried messing with the CHMOD settings of the files but no joy. My manager.conf looks like: ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = secret ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.73/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user I can successfully telnet into the manager interface through shell on the local machine and winxp machine on the LAN I have moved asterisk.reload into /bin, and if I run it from the shell I get a successful? Output: pbx01:~# /bin/asterisk.reload Asterisk Call Manager/1.0 Can anyone help? It is the same error the online example gives. Is it something to do with specific admin rights in xorcom, or have I missed something fundamentally wrong out? I have checked the php files and the paths seem to be OK (default * installs) I have a couple of ideas as to the problem: -PHP needs something enabled e.g safemode? -Xorcom has changed something phpconfig needs e.g * not running as root or something? Many Thanks, C ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Thanks for the chmod, it definitely needed that! I didn't have to change the etc.sudoers file though. I'm running Debian, via the great Xorcom rapid installation. I didn't change the permit lines either as this is just attesting box and im not worried about security. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hecken, Guido Sent: 25 February 2005 14:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = secret ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.73/255.255.255.0 Do this in manger.conf, where xxx.xxx.xxx.0 represents your network: [admin] secret = secret permit=xxx.xxx.xxx.0/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user To get the reload script running, you must allow apache (or the account apache runs under) to execute scripts. Therefore use visudo (in Fedora) to add the following line in /etc/sudoers apache ALL=(ALL)NOPASSWD: ALL To write the config files in /etc/asterisk with phpconfig.php you need to give apache the rights to do so. A simple chmod -R a+w /etc/asterisk should do the job. I know, there are more secure methods to do this, but it works for us. Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Hi, Thanks for the batchfile type, it's a handy one. I'm not editing over the internet, just local LAN for testing ATM. Protected via firewall. Would it not be fairly secure using a combination of the following: .htaccess file VPN? https access? Limit apache to only allow certain IP's? And the public keys thing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 25 February 2005 15:49 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? On Fri, Feb 25, 2005 at 01:52:21PM -, C. Tomlinson wrote: Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? Basically you need to run one shell command. In linux I'd use: ssh [EMAIL PROTECTED] asterisk -rx reload As this is a platform without native support of ssh, you can use the command plink to get basically the same effect. Create a putty configuration called rapidroot to connect to [EMAIL PROTECTED] and use something like plink rapidroot asterisk -rx reload in a batch file. Or use [open]ssh from cygwin, if you're more comfortable with it. You should use public-keys authentication to get better control . Actually you can configure a certain public key so it will only allow running one single command (asterisk -rx reload, in your case). This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. OVER THE INTERNET??? See my recent post on the previous thread about phpconfig. Allowing phpconfig to do the same is quite insecure. Also consider using mc from the shell. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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
