[asterisk-users] Re: Correct latency values in sip show peers
Eric ManxPower Wieling wrote: The times shown are the time to get a response to a SIP OPTIONS packet sent to the phone, not the time to get a response from an ICMP ECHO (ping) packet. What's the difference between yours and mine mail? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: System from AMI
Richard Lyman wrote: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: dummy Exten: 2 Priority: 1 In extensions.conf [dummy] Exten = _X,1,System(*some command*) remember your permissions OK, thank you! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Correct latency values in sip show peers
Rolz wrote: I was wondering if anyone knows how accurate the values are when you do a sip show peers from the CLI. My configuration is: Asterisk box (192.168.1.102) - gigabit switch - PC running x-lite (192.168.1.100) the CLI reports 101 ms delay however, ping is showing 1ms delay Where is the extra 100ms coming from? The softphone response? I'm not sure, but I think that ping isn't OSI ISO Layer 7, while softphone is. So, this 100 ms can come from there. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: System from AMI
Richard Lyman wrote: if you are unable to get '!' to work... there are other ways of doing this. manager originate can do this, use a local channel and point it at either a context/exten with an echo/system call, You mean like this? [testdelete] exten = myexten,1,System(rm /tmp/test.txt) exten = myexten,2,Hangup Action: Originate Channel: SIP/987987 Context: testdelete Exten: myexten Priority: 1 Callerid: 987987 Timeout: 3 ActionID: uniqueID Can I use this in production use? I need to execute this up to 1000 times a day (in period of few hours). or with the application/data method. Can you please show me some example? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: System from AMI
Richard Lyman wrote: Action: Originate Application: System Data: /path/to/script Channel: Local/[EMAIL PROTECTED] Context: dummy Exten: 2 Priority: 1 In extensions.conf [dummy] Exten = _X,1,Wait(2) Exten = _X,2,NoOp fyi: manager originate is channel + context + exten + priority OR channel + application + data not both. So, you are saying that this should look like this? Action: Originate Channel: Local/[EMAIL PROTECTED] Application: System Data: /path/to/script And this is all that's needed? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Is it possible to install CCM on a Linux platform ?
Olivier wrote: Hi, I know this question doesn't exactly relate to the core of this list but I thought it does relate to its hacker spirit. Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance ? A friend of mine working for a Cisco VAR told me his colleagues couldn't make it, even for testing purpose. Do you agree ? Regards This really isn't Asterisk users question. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How is this feature called ?
Olivier wrote: No, I'm far from inventing features, yet ! ;-) It's a feature offered by Alcatel and I wanted to find in documentation, a way to reproduce it, just in case I'm asked to do so. I think it's the equivalent of call screening, but from caller perspective. Cheers Well I don't like this option at all. User should have ability to send incoming call wherever he likes to, and nobody should override that. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: System from AMI
Lee Jenkins wrote: You have to login into the AMI server with proper credentials and send commands. Hi Lee! Thank you for your mail. I do login to the Asterisk 5038 port and I have all credentials. What I'm asking is what Action allows me to execute system command. What I tried is this: Action: Command Command: ! ls Response: Follows Privilege: Command --END COMMAND-- OK, maybe he doesn't show output, so I have tried this: Action: Command Command: ! rm /tmp/test.txt Response: Follows Privilege: Command --END COMMAND-- But this doesn't work neither (tmp/test.txt is stil lthere). Is there any solution to this problem? I wrote an AMI test application a little while back. It gives you the ability to login into the AMI, send commands and snoop packets being send out. Great way to get familiar with AMI commands and packet structure. http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx You'll see the download under Manager API Testing Utility. It's freeware. I don't believe that your application can help me with this one. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How is this feature called ?
Olivier wrote: Hi, Your colleague has forwarded his incoming calls to his secretary. How do you call the feature allowing you to circumvent your colleague call forward to make your colleague's phone ringing ? Hi Oliver, is this some new feature that you have invented and you need to come up with some name for it? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP Video Camera
KokMengLoh wrote: Hi, Does anyone know of a Video Camera that is based on SIP? There are lots of Video Phones out there, but I can't seem to find a Video Camera. What would you do with SIP video camera? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] System from AMI
How to execute some system command from AMI? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: AOC billing
Stefano Corsi wrote: is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Ciao Stefano! Since I'm not a programmer I'm waiting for some AOC solution, but nothing was developed so far. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: just on my LAN
Josu Lazkano Lete wrote: hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz Zaptel 1.2.16 http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz Libpri 1.2.4 http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz Addons 1.2.5 http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.5..tar.gz Sounds 1.2.1 http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1..tar.gz You realy should read this http://www.voip-info.org/wiki-Asterisk -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI - delete voicemail
How can I delete voice mails (all, new and old) from AMI? I thought that I could use Action Command, but there is no command to delete voicemail. So I figure it out to use system command and execute rm /var/spool/asterisk/voicemail/default/100/INBOX/* and rm /var/spool/asterisk/voicemail/default/100/tmp* but I don't know how to do that from AMI. Any suggestions how to do this are welcome. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SRTP testers needed
marek cervenka wrote: please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,gxp-2000, minisip, twikle, ) Does it work between two asterisk? If I use gxp-2000 = * = * = phone that doesn't support SRTP will it work? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: wrong values in duration and billsec in CDR
C F wrote: So, how to solve this problem? Get an ISDN line, or maybe just VoIP. This really isn't answer to my question ;) Why not? FXO is answered as soon as you go off hook. There is no real way it will work on FXO, unless you get an ISDN or all VoIP lines. C F, thank you for your mails. You helped me to understand what's the problem. In joke I have mention that you didn't answer to my question (because it's impossible to answer), but you have helped me to overcome the problem (but not to solve it). Regards, -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: wct4xxp problem
Steve Totaro wrote: All providers will do this if a line is in alarm for a couple days. The FIRST thing to do is call them and make sure it is turned up. Once and only once was I called prior to the CO/NOC turning down a circuit that had been in alarm for two days straight. Yes, and before they turn down the line they should call you and check what's wrong. What if that was your backup line and you haven't noticed that it's in red alarm? Then one day when you need that line it doesn't work. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 302 Moved temporarely
Eric ManxPower Wieling wrote: I believe that if Asterisk receives a 302 back from a device when sending a call to it (i.e. the SIP device has Call Forwarding set) Asterisk will do the right thing. This may only work for redirects to locations on the same Asterisk server. This is NOT the case with registrations, only call setup. Well, this happens when I try to make outgoing call through Voipbuster SIP provider. I register fine, but sometimes when I try to call someone I receive that message (probably they redirect me to some server with lower usage) and I can't establish phone call. But, as far as I have understand you, this should work. Right? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: wrong values in duration and billsec in CDR
C F wrote: Zap channels on FXO are considered answers as soon as Asterisk finished dialing. So, how to solve this problem? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: About Pickup Grandstream
LKS GMAIL wrote: My question is that it’s impossible to pick up a call from ZAP, IAX or mISDN with my Ext Key of my GrandStream. Yes, it's possible. Read documentation and configuration examples on Grandstream Internet pages. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: wct4xxp problem
Tim Panton wrote: I once spent a week struggling with this sort of symptom to find in the end that the ops guys had got fed up with my line being in 'alarm' on their console and disabled it at their end. One phone call later it was re-enabled ! Have you changed your provider after that? If you haven't then you don't deserve better. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: strange ring
Pryakhin Dimitry wrote: Hello Im having strange asterisk ring. I'm dialing PSTN network, then I get my call answered and I hear a person talking but the same time remote person can't hear me. They get a ring tone. What can be the problem? Where do I need to look for it? Do you have r in your Dial command? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 302 Moved temporarely
Does Asterisk supports 302 SIP message - Moved temporarely? I have found mail from Olle E. Johansson (April 2006) that Asterisk doesn't support redirects of registrations. Does it support now? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: wrong values in duration and billsec in CDR
C F wrote: So, how to solve this problem? Get an ISDN line, or maybe just VoIP. This really isn't answer to my question ;) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: strange ring
Pryakhin Dimitry wrote: But the thing is, that asterisk playin ringtone to a called person like that persone is calling some one. Please send the complete Dial command. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 302 Moved temporarely
Kevin P. Fleming wrote: No. If you ask Asterisk to register to a SIP server, and it receives a 302 back from that server, it will not re-attempt to register to the redirected location. Are there any plans to implement this feature? Any work done allready? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Zaptel 1.2.16 Released
Asterisk Development Team wrote: The Asterisk and Zaptel development teams have released Zaptel version 1.2.16. On http://www.asterisk.org/downloads there is still link to 1.2.15 -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Call counter for sip misbehaving
Rizwan Hisham wrote: Im using asterisk1.4.0 . declaring type=peer solves the problem. but if anybody knows why its not working for type=friend, plz share. Please try 1.4.1, this should be fixed. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: While the VoIP-Info.org site is down...
Matt wrote: Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in Central PA on a server with RAID. Additionally, it is at the end of 95Megabytes/second on a BGP redundant connection. Please feel free to use it, if the community feels it can be useful... additionally, I would love to setup some rsync mirrors with others so that we can have redundant backups of this very valuable information. I don't think that we need new wiki. It would be better to wait that old wiki comes back and then make two mirror sites. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: While the VoIP-Info.org site is down...
Stephen Bosch wrote: RAID or no RAID, the site should have one or more mirrors. As soon as wiki comes back, mirrors should be created. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: voip-info.org status update
James H Thompson wrote: I will definately be looking for an easy way to create a mirror site once voip-info.org is back up. This is made difficult by the dynamic nature of the site, but its been on my list of things to do for a while now. Hopefully this will happen before next crash :) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7912
Matt Putnam wrote: I have 3 cisco 7912 that all stoped working at the same time on sunday. There is nothing on the display and the menu and hold buttons are lit. Resteing produces the same results the phone dosent respond. Anyone have an idea how to fix this or if it can even be fixed. Ive done some searching online and havent found anything useful any sugestions? Same thing happened to my two phones. Luckily they were under warranty so I sent them back and I have received new phones. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What happend to voip-info?
Gordon Henderson wrote: The site is pingable, so I'd suggest it's either crashed in some awkward way and just needs resetting, but you never know... Voip-info.org is down due to a hardware failure. Will be back soon. Thanks for using voip-info.org! [EMAIL PROTECTED] -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk on mini-itx
Gordon Henderson wrote: I've built several systems based on this motherboard (the 1GHz fanless one) Compressed codecs are fine - as long as you aren't transcoding ;-) I figured I could push 30 non transcoded calls through one, but I've never had the ability to fully test it out. The max. I had going on one system was 20 calls. What was CPU usage during this 20 calls? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: AMI - DBPut
Lee Jenkins wrote: Try putting quotes around the value. I played with it a while back only a little, but I can't remember if quotes did it or I ended up having stripping the quotes off myself when I retrieved the value ... My first mail was copy/paste, so I'm positive I didn't make any error typing. Now I have tried again with the similar input and it works. Action: DBPut Family: checkin Key: 319 Val: yes Response: Success Message: Updated database successfully I really don't know why it didn't work yesterday. Has anybody head similar error/problem? Now I wonder, is it stable enough for production use? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Number of SIP messages per minute
Mark Davies wrote: I’ve just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me. Put Ethereal and count :) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 1.4 compile issue
Wai Wu wrote: I am use Fedora 3, and run into a 1.4 compile issue. I recommend you to start using Cent OS 4.4 - it's basically RHEL. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI - DBPut
I'm using AMI on Asterisk 1.2.15 and I'm having problems with DBPut action. If I execute this: Action: DBPut Family: checkin Key: 316 Val: yes Response: Error Message: Missing action in request I don't put anything in Asterisk DB. If I execute this: Action: DBPut Family: checkin Key: 316 Val: yes Response: Success Message: Updated database successfully Then I put data in Asterisk DB, but that data has and . How to enter data in Asterisk DB without this brackets. fc4*CLI database show /checkin/316 : yes /dozvola/148 : yes -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: queue information into db
nik600 wrote: new update 11/03/2006 - added the module stats - updated the file db.sql with sql instructions for the creation of queue_stats table - added the files view.sql I'm in no position to test your product now. Hopefully I will find some time soon. Please keep group informed about new updates. Bye, -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk
C F wrote: Tomislav, really? and how does it show up on my POTS line? It only can be seen if other end is also on Optima provider. Ant it is shown exactly as originator has define it. It's strange when you, for the first time, get the phone call from unknown number and you see his name at your display :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
Axel Thimm wrote: As fast as they read asterisk-announce ;) I doubt that you are that fast ;) but I thank you for answer. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Help: CallerID Name not being sent on outbound PRI trunk
Matt wrote: Thanks I was just about to say this. You CAN'T send caller-id-name. To be able to set name you need to set it with Telcordia or whomever manages numbers in your country. Optima provider in Croatia allows users to set up CallerID name on outgoing PRI calls. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: visdn, misdn and the hell
Massimo Nuvoli wrote: I think the ISDN part of asterisk is very important, in Italy there is a lot of equipments that are ISDN and not ANALOGIC or PRI, and with no ISDN stable support it is impossibile to port asterisk on the real world. In Croatia also. Small companies are just to small for PRI and they all use ISDN BRI lines. Wath i see now is that a lot of integrators are doing this: using external box to avoid at 100% the isdn problem in asterisk. Very bad, we go to use proprietary designed hardware and software, external components, more complexity, more point of failure. Definitely agree with you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Faxing Support
Andrew Kohlsmith wrote: Undue? Digium requires disclaimers so they can dual-license it for ABE and other commercial vendors. You're purposely twisting and distorting the reality with these weasel words. I understand Digium strategy but I don't agree with it. I think it's wrong not to include code in Asterisk just because they won't be able to use it in ABE, so noncommercial version would be better. Asterisk isn't strong because of ABE and commercial installations, but because of big number of users and developers. Doing thing's like this Digium is pushing people away from Asterisk. If you don't like it, use something else. There's no need to take jabs at the company. You are not helping neither. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Digium cards on Vmware
Kevin P. Fleming wrote: The card manufacturer is irrelevant, as is the type of card. VMware does not currently provide any sort of PCI bus passthrough to virtual machines.. Hopefully this will change soon. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: FAX using T38
Steve Underwood wrote: I'll do it for 30% less than they quote. :-) I didn't see on their pages, what is their price? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Micros-Fidelio - billing in hotel
There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something that is used world wide. Any informations are welcome. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
Axel Thimm wrote: Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. They don't have 1.2.x version there? How fast do they make package since source version is out? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: build rpm fails
Tomislav Parcina wrote: They don't have 1.2.x version there? Newer mind, I found it :) How fast do they make package since source version is out? This question still stands. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Registrations, how many is too many?
voiplist wrote: We do not use dyndns for anything, not sure what we would even use it for. We do have lots of hostnames to different systems in our sip.conf, I have changed them all to IP to see if this helps. So, you think that maybe when DNS gets hosed up that it could cause SIP to just tank on a high volume system? Of course you need to have DNS server installed on Asterisk machine. So, Asterisk will ask that machine for DNS records. If that machine doesn't know the answer (because Internet connection is down), at least Asterisk will get fast answer so it won't stop responding. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 1.4 lost internet internal phones loose registration
Thomas Kenyon wrote: Asterisk also seems to barf if it makes a registration/renewal request and it doesn't receive a reply in a timely fashion which will obviously happen if your internet connection disappears. (all versions I've used). That's why people should use dnsmasq. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED
Kristian Kielhofner wrote: Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Hi Kristian! Thank you for your work. I'm not able to test this right now, but I'll sourly need this sometimes. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: queue information into db
nik600 wrote: i'm sorry but due to some problem the software will be released not first than Wednesday 7/02/2007. i'll post a message . This should be Wednesday 7/3/2007. right? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Sending SMS
Supa wrote: Try this: http://www.bayhamsystems.com/asterisk.html Works for me just fine, and it is very easy to get up and running, even with older version 1.2.3 I don't see a point of using providers as Bayhamsystems. First, it's unpractical to send SMS from phone. If I'm going to use web interface, then is better to use some provider that has web interface just for that (or maybe they will provide application to send messages to groups or in certain time). Only reason why I would like to do it true Asterisk is if I could use my VoIP or E1 provider so that I get only one bill. But using Bayhamsystems that isn't a case. So, why people use such providers? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: queue information into db
nik600 wrote: actually it isnìt released under any type of licence. if you want i can put the code on my web site (but no earlier than the next week) Please do. And it wouldn't hurt if you, somewhere on the page, put that is released under GPL or something similar. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: fax support
Olle E Johansson wrote: However, the 1.4.0 release is buggy, so either use 1.4 from subversion or wait for 1.4.1. Have you put this information somewhere on web page of Asterisk? I think its fair enough to say - look, this doesn't work as it should, use 1.2X or 1.4 from subversion. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP interface status and calllimit
James Fromm wrote: I've reviewed the bugs reports. I didn't see anything that applied to this. Have you? Could you point it out to me? Just for the record, I believe this is what you are looking for. http://bugs.digium.com/view.php?id=8800 -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Revolution Call Accounting Desktop
Has anyone used this billing application with Asterisk? I have one potential costumer (hotel) that will use application that connects with Fidelio/Micros and so they can use Revolution Call Accounting Desktop for billing. More info about product you can find on this page http://www.telecost.com/revcall.htm Now, has anybody implemented this with Asterisk? Any known issues? Is it ready for production system? I would appreciated any info about this. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: queue information into db
nik600 wrote: In the last months i've developed a web application for the use of an asterisk call center. Yuo can - make calls from a web interface - login/logout in queue - view members logged in a queue - view callers queued in a queue - pickup a callers from a queue What is license of this application? Can it be downloaded from somewhere? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk sip and radius authentication
Hi Sergio! If you make it work. Please send some feedback to the list. Tomislav From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Inigo IbánezSent: 8. oujak 2006 15:39To: [EMAIL PROTECTED]; asterisk-users@lists.digium.comCc: 'Sergio Iñigo Ibáñez'Subject: [Asterisk-Users] Asterisk sip and radius authentication Hello all, I am new in asterisk configuration. I want to configure a Radius server to authenticate the sip users of asterisk. I have trying to use the next document: http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html Can you help me? Regards, Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] news-reading question
Hi Dan! Yes, that news group follows this mailing list. They head some problem in past few days. Now it's working. Tomislav From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan MillerSent: 9. oujak 2006 20:20To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] news-reading question Is there some way I can follow this list from a newsgroup?? Is this the same as the gmane group gmane.comp.telephony.pbx.asterisk.user ?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: 9. ozujak 2006 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr data
Hi Aleksander! Can I setup accountcode in sip.conf or I need to set it up in extensions.conf? is this right exten = s,1,Set(CDR(accountcode)=$CALLERID(number)) or exten = s,1,Set(CDR(accountcode)=$CALLERID(name)) Tomislav From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: 9. oujak 2006 16:33To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] cdr data That is what the accountcode field is for, you can set a unique accountcode for each devcice if you want to. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ON DEMAND call Recording
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: 8. ozujak 2006 23:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording I would find two possibilities: 1. on demand. Dial another extension number after the call, what executes a system command 2. automatically. Add in the dialplan the system command after hanging up. Hi Ronald! The second option is weary interesting. I think I have enough knowledge to make it work. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ON DEMAND call Recording
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: 7. ozujak 2006 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote: ya i found it it *1 to start recording from the caller end Also pushing *1 again stops recording. Do you know how to send that recording to e-mail address that is specified in voicemail.conf? That will be a real cool option. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Random Disconnects
In article 77758c190601240743o3ae310dbi28b2f79a93965776 @mail.gmail.com, [EMAIL PROTECTED] says... I am not very satisfied with this, though. I want to use some features (like Park) that apparently don't work well with reinvites. Have any of the rest of you had any luck troubleshooting this problem? Your RTP stream doesn't pass thrue Asterisk and it can't hear that you have pressed any key (that you are requesting that he parks the call). -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: uip200 transfer calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... All, - 1 call transfer - Call comes in for uip200. can trasnfer it just fine. - 2 call transfer - Call comes in - then a second call comes in I can longer transfer either call? I can toggle between them but not transfer. Does anyone know how to acomplish this? Are you trying attendent or blind transfer? If you are trying att transfer, than that should be your 3rd call and maybe your phone dosn't allow it. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: RE: RE: RE: Spandsp
I have included logger.conf and now I see that problem is with loading libspandsp.so. I dont have that file in /usr/src/asterisk-1.2.1/apps Can you tell me where do you have it? Does it means that spandsp wasn't installed corectly? This is what I get when I try to start * with logger.conf. [app_txfax.so]Jan 16 10:01:35 WARNING[7933]: loader.c:325 __load_resource: libs pandsp.so.0: cannot open shared object file: No such file or directory Jan 16 10:01:35 WARNING[7933]: loader.c:325 __load_resource: libspandsp.so.0: ca nnot open shared object file: No such file or directory Jan 16 10:01:35 WARNING[7933]: loader.c:554 load_modules: Loading module app_txf ax.so failed! Jan 16 10:01:35 WARNING[7933]: loader.c:554 load_modules: Loading module app_txf ax.so failed! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Mediatrix windows-based setup?
Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE: RE: RE: Spandsp
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Download hylafax, and iaxmodem. Set up a friend extension as iax, and let it rip... it's a slam dunk. I think I have found the real problem source (spandsp, not txfax) and maybe now I solve it. If I don't manage, I will surtnely lisen your suggest. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Spandsp
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Do you have the spandsp libraries in your library path?, by default they go into /usr/local/lib In that dir I have libspandsp.a, libspandsp.la, libspandsp.so (softlink), libspandsp.so.0 (softlink) and libspandsp.so.0.0.1. Is that all i need to have? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Spandsp
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... makefile.patch is buggy. Compile app_rxfax and app_txfax by hand. I have tried two patches, one was applied without returning any error. So I gess makefile was pattched OK. Anyway, I'll try to patch it by hand. First I need to find instructions (yesterday I have find them on some web pages, hopefully I will be able to find them today). And I have one question. I need to copy old unpached MakeFile from source tar file, right? P.S. Do you have link to instructions how to compile app_rxfax and app_txfax by hand? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Spandsp
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Yup, do you have /usr/local/lib listed in your /etc/ld.so.conf ? , you may also need to run ldconfig after compiling spandsp, but before compiling rxfax and txfax. No I didn't have /usr/local/lib in /etc/ld.so.conf. Now I have added this line /usr/local/lib and then rebuild Asterisk but there is the same problem. From which dir do I run ldconfig? I have run it from /sbin/ but I didn't recive any output on console. Should it be that way? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: RE: Spandsp
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
I solved with this simple makefile:
all: app_rxfax.so app_txfax.so
app_rxfax.so: app_rxfax.c
gcc -shared -Xlinker -x -O2 -D_GNU_SOURCE -Iinclude -I../include -o
$@ app_rxfax.c -lspandsp -ltiff
app_txfax.so: app_txfax.c
gcc -shared -Xlinker -x -O2 -D_GNU_SOURCE -Iinclude -I../include -o
$@ app_txfax.c -lspandsp -ltiff
but as usual cutpaste is problematic.
When I try to patch with this one I get error message. For me, this
patch seams to work (I don't get error message).
I have send mail to Steve Underwood, hopefully he will be able to help
me.
--- Makefile.orig 2006-01-11 18:39:21.0 +0800
+++ Makefile2006-01-11 18:40:46.0 +0800
@@ -52,10 +52,14 @@
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h
$(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h
$(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),)
+APPS+=app_rxfax.so app_txfax.so
+endif
+
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L
$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
@@ -100,10 +104,16 @@
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} $(CURLLIBS)
+app_rxfax.so : app_rxfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
+app_txfax.so : app_txfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} -
L/usr/local/pgsql/lib -lpq
--
Tomislav Parcina
[EMAIL PROTECTED]
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[Asterisk-Users] RE: Fax RX and SIP/IAX
In article 1591CC66BC72C847A8384D7D43065EEB31D729 @st_server2.solacomm.com, [EMAIL PROTECTED] says... Spandsp app_rxfax / app_txfax will work over sip and IAX Currently in use here... http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax Do you know what could be the problem with this one? Copy of mail that I have send to Steve Underwood. On Fedora Core 4 I have installed * 1.2.1 with zaptel, add-ons and sounds. On FC4 I have installed libtiff 3.7.1, libtiff-devel, libxml2- 2.6.19 and libxml2-devel. In /etc/ld.so.conf file I have put line /usr/local/lib And in /usr/local/lib I have those files (libspandsp.a, libspandsp.la, libspandsp.so (softlink), libspandsp.so.0 (softlink) and libspandsp.so.0.0.1). I have copied and untar spandsp 0.0.2pre22 in /usr/src/. Than I have copied app_rxfax.c, app_txfax.c and apps_Makefile.patch in /usr/src/asterisk/apps/ dir. I have execute patch file (in attachment) and I didn't get any error on console. Then I have make clean; make; make install - reinstall of Asterisk. Then, then I try to start * I get this on console. [cdr_custom.so] = (Customizable Comma Separated Values CDR Backend) [format_ilbc.so] = (Raw iLBC data) == Registered file format iLBC, extension(s) ilbc [app_curl.so] = (Load external URL) == Registered custom function CURL == Registered application 'Curl' [EMAIL PROTECTED] apps]# And Asterisk doesn't start. Can you please tell me 1. how to check what is the problem? (my guess is that app_txfax.so wasn't installed like it should) 2. why did I get this problem? (wrong applied patch? But I didn't get any error message!) 3. how to solve this? I really hope that you can help me. Thank you for your time! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: AoC (Advice of Charge)
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Is this on bugs.digium.com ? I got on some other things and this isn't my priority at the moment so I didn't check. If you check and find out please send me the link. Soon, I'll have to come back to this one. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: automon - one touch record
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It shouldn't make a difference, but should not and does not isn't always the same thing! We can't discus about this topic. It is simply meather of opinion. You think that is important and I don't. I like to be thorough and systematic when problem solving... Me to, that why I dont bother with erelevant things and care only about things that are relevant. Like I said before, it is mine and your opinion. It has no point discusing about it. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: RE: RE: Spandsp
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... This is not a patch is a simple makefile! Save it in a file, with correct tabs, and run: make -f file in the dir where app_rxfax.c resides. :)) I have done that (I needed to change a file several times). It created app_rxfax.so and app_txfax.so. But again I'm unable to start *. Can you please send me that file like attachment to [EMAIL PROTECTED] maybe I didn't copy/paste it right. P.S. Is there any other way to recive fax and not use spandsp (app_rxfax app_txfax? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: R: app_rxfax.so and app_txfax.so
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It works. Lucky you! I wasted hours trying to make ti work and I allways end with this [cdr_custom.so] = (Customizable Comma Separated Values CDR Backend) [format_ilbc.so] = (Raw iLBC data) == Registered file format iLBC, extension(s) ilbc [app_curl.so] = (Load external URL) == Registered custom function CURL == Registered application 'Curl' [EMAIL PROTECTED] apps]# Trying to start Asterisk... -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Easy to Access Telephone Directory AGI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Really interesting. Thanks Hannes!! Hear, hear! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE : Re: RE : codecs order and so on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Calling zap = no problem, Ulaw is choosen Calling pstn provider =fail (I need g729 but Ulaw is choosen) Call from zap = no problem Ulaw is choosen Call from pstn = no problem g729 used... When you call out * establishes two channels. One is between Ua and *, and another between * and Zap (or provider). If you call out, asterisk first negotiate codec for that channel. Then it tries to nagotiate codec for second channel. When you call your provider it can't nagotiate because he doesn't have g729 codec. This is reason why you have problem, and I have explain how to solw it. There is nothing else I can say to help you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Recommend Fax Hardware for T1 PRI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Wonderful advice. Both of these solutions actually fail for most people. Digium card worked for me. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp
I have tried to install spandsp. On fresh installed FC4 and Asterisk
1.2.1 with zaptel, addons and sounds. I have libtiff, libtiff-devel,
libxml2 and libxml2-devel RPMs installed.
I have untar spandsp-0.0.2pre22.tar.tar and have run
./configure
make
make install
then I have execute patch (at the end of mail) and I didn't recive any
error.
I have again run in /usr/src/asterisk-1.2.1/ dir
make clean; make; make install
and when I tried to start *, it fails when tries to load app_txfax.so.
What could be wrong?
[format_ilbc.so] = (Raw iLBC data)
== Registered file format iLBC, extension(s) ilbc
[app_curl.so] = (Load external URL)
== Registered custom function CURL
== Registered application 'Curl'
[EMAIL PROTECTED] /]#
*** patch file ***
--- Makefile.orig 2006-01-11 18:39:21.0 +0800
+++ Makefile2006-01-11 18:40:46.0 +0800
@@ -52,10 +52,14 @@
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h
$(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h
$(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),)
+APPS+=app_rxfax.so app_txfax.so
+endif
+
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L
$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
@@ -100,10 +104,16 @@
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} $(CURLLIBS)
+app_rxfax.so : app_rxfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
+app_txfax.so : app_txfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} -
L/usr/local/pgsql/lib -lpq
--
Tomislav Parcina
[EMAIL PROTECTED]
___
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[Asterisk-Users] Re: automon - one touch record
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Also: What are the SIP CanReinvite settings for these phones? This shuldn't be important because he have w and W in his dial plan. * doesn't allow reinvite if you have t, T, w or W. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Wake-Up Call
In article [EMAIL PROTECTED] ny.censys.net, [EMAIL PROTECTED] says... Something to think about is this too, when completed scheduling, ask would you like to notify another extension, so if the first does not answer in two attempts, ring a cell phone or such. But I cannot complain, I use the wakeup call function every day, and it is definitely better than any alarm clock or pbx reminder available. Yes, I like it. It could have more features, but I won't complain ;)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: mpg123 removal
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... How can you convert mp3 to gsm? mencoder? Do you have an example? You can use this page. http://www.asteriskguru.com/tools/audio_conversion.php -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Re: Remotely reboot SIP Phones ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Figured it out :) Basically, you have to have a file called syncinfo.xml in the tftp root directory, with the following contents: SYNCINFO IMAGE VERSION=* SYNC=1/ /SYNCINFO Also, in SIPDefault.cnf or the phone's configuration file, stick: sync: 0 somewhere so the phone's sync value doesn't match the value in syncinfo.xml. If you make a change of sorts, just run sip notify reboot-cisco username at any time in asterisk and it'll send the notify to the phone. If the phone is in use, it waits until it's idle, once it is, it waits 20 seconds and then checks the syncinfo.xml file, and if the values of sync are different, it reboots :) Hi Aron! What Cisco phone do you use? I use 7940 with SIP firmware version POS3- 07-5-00. For me it works but on wery strange, I shuld say wrong way. I have put syncinfo.xml in tftp root and when I enter this in * CLI pbx*CLI sip notify reboot-cisco 201 202 Unable to find notify type 'reboot-cisco' pbx*CLI sip notify cisco-check-cfg 201 202 Sending NOTIFY of type 'cisco-check-cfg' to '201' Sending NOTIFY of type 'cisco-check-cfg' to '202' Like you said, after 20s it looks for two files in tftp root dir - dialplan.xml (why?) and syncinfo.xml. Then Cisco waits. I have wait for more then 12 min and nothing happened. Then when I decaided to pick up handset, then it started to reboot. He reboots for 2-3 min. If my boss needs to make a important phone call I'll get fired :)) Why he vaits that I pick up handset (or press any bottun)? Anyway, thank you for this one (if I don't get fired :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE : codecs order and so on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In extensions.conf, I change the context for each call, Asterisk doesn't care of codecs in contexts, it uses the order of general... Could be good to have Ssterisk making a match between codecs in General and the context used to make a call But definitiely, Asterisk choose g729 even if I am in the zap context Any idea, help is welcome. Phones usualy use only one prefered codec. So, if your phone supports ulaw and g729, it will use only one of those two to communicate with *. Once the phone is authenticated with * he allways use the same codec. So you have to get use that on that side is that specific codec. What is on another side (SIP, Zap, IAX2...) and what codec other side uses, determinates do you need codec translation in * box. If you need codec translation then you need to have licence (for g729). I hope I have make it clear for you. Solution: Count do you get more outside ulaw or g729 calls (at the same time). If you get more ulaw calls then use ulaw codec on SIP phones. Buy the same number of g729 licences as you need simultanius phone calls to that provider. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer sounds - notifications
When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear transfer. I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear anything (that person waits for me to say something) and I don't know that I'm back to transfered person. I hope that I have make it clear enough. Anyway, how can I solve this one? I would like to hear that the phone of extension is ringing, and I would like to konw when I'm speaking again with my caller. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Transfer sounds - notifications
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... On http://www.voip-info.org/wiki-Asterisk+config+features.conf: ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer You could try these to see if that makes a difference?... Thank you. I have uncommented those and restart asterisk but it is the same. I hear beep only when I establish att transfer and other party doesn't want to take over a call. So, other party hangs up before I do, and in that case I hear beep. In all other cases I don't hear any tone. I couldn't done anything wrong?!? Do I need to add any DYNAMIC_FEATURES in extensions.conf? This is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 featuredigittimeout = 1000 ; Max time (ms) between digits for feature activation. Default is 500 [featuremap] blindxfer = #1 ; Blind transfer ;disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer [applicationmap] ;testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkeys to ;callee if #9 was pressed -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Recommend Fax Hardware for T1 PRI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I assume I need a TDM400P (TDM20B flavor for 2 analog stations), but I am not sure. You can buy ATA (analog terminal adapter) or the card you mention. Bouth of them shuld work just fine. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Failover Device?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. I'm sure that I'm not the only person that has notice that there is lots of people that start new thread by replaying to old message. That way neither them, or lots of other people, sees that mail as new therad. The truth is out there! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Remotely reboot SIP Phones ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Yeah, that should theoretically work, but I've got about 60 cisco phones that don't respond to the check-sync. If you ever make it work, please anounce it on the group. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk CLI | more
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... If you're wanting to scroll through output from a CLI command, use: asterisk -rx command | less Thank to bouth of you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Remotely reboot SIP Phones ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... An example SIP friend is defined as [112], so we could now type, from the CLI: sip notify polycom-check-cfg 112 sip notify cisco-check-cfg 214 doesn't seam to do anything. I have sip_notify.conf in my /etc/asterisk/ directory. Cisco 7905 and 7940 phones don't react on that command. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wake-Up Call
I have setup wake up call in * following those instructions http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP and it works fine. Now I have few questions. - When I arrange wake up call, does it call me only that day or I can set it up for whoole week? - Can I set it up for some other extension or only for one I'm calling? - Can this AM, PM be in 24h format? That is all (for now :)). -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Call logging
In article 6A1C243A7E2E824293FABC3042045790930851 @dtw_localmail.strtrade.com, [EMAIL PROTECTED] says... Hello all, is anyone aware of any open source call accounting software for Asterisk? Something that can parse out Asterisk's call detail records and generate on-demand reports? Check out Asterisk-Stat: CDR Analyser http://areski.net/asterisk-stat-v2/about.php?s=0 -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Start recording after call started
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Try experimenting with this: [general] featuredigittimeout = 1000 ; Max time (ms) between digits for ; feature activation. Default is 500 It seams it works. Thank you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Re: Ominiis Asterisk TAPI driver
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... CounterPath's X-Pro Tapi softphone has this I think? http://www.xten.com/index.php?menu=X-Series (select the EU region) I think they have a trial...downloading it now. Thank you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP security
In article [EMAIL PROTECTED], trixter@ 0xdecafbad.com says... to add to this, given the state of MD5 and its 'security' or lack thereof, its a bit over simplistic to just say md5 without adding that its actually 3 md5 hashes... Precomputing is harder (but not impossible) because of the way its done. The nonce makes it a little harder - although the nonce is known even by an attacker ... To make long story short, SIP can be cracked (like evrything else). It isn't so simple like sniffing the line because data is encripted. I don't have to put anything extra in my sip.conf (or any other conf file) or in my softphone for basic security (md5 encription), because all is allready there. Have I got that right? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
