Re: [asterisk-users] [Zaptel] Why no port to Windos?
Which has existed, in one form or another, for years. I was using a voice enabled faxmodem a decade ago to answer my phone. The software that came with it (don't remember the name, but WinFax also does/did this) even allowed for a simple IVR, for mailbox selection and whatnot. The only things it didn't do that asterisk does (and would be useful to the average Joe) was support multiple phones/extensions and send voicemail messages via email. I think what you are looking for is named SuperVoice : http://www.supervoice.com/asp/products_supervoice_fax_products.asp I was using it to receive faxes and voicemail. It didn't email me my fax and/or voicemail but it would page me the number of faxes and voicemail I had everytime it received one or the other. The only problem I was having was that, since running on Win9x, sometime my phone line would stay busy and that would signal the time to go home and reboot the computer. Maybe some people would like Asterisk for windows, but I would not touch it with a ten foot pole :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Directory Format
A little off topic (sorry..:) ) but anyone know what format Cisco phones use for their contact dirctories. I want to set up my contact lists on the phone, and cannot seem to get any info on it. I am working with a 7970 on Asterisk 1.4.8. 7940 and 7960 use this format of XML file (probably the same on 7970) CiscoIPPhoneDirectory TitleEmployee directory/Title PromptOpen Source Rock/Prompt DirectoryEntry NameEmployee A/Name Telephone7001/Telephone /DirectoryEntry DirectoryEntry NameEmployee B/Name Telephone7002/Telephone /DirectoryEntry /CiscoIPPhoneDirectory Check also Open 79XX XML Directory : http://web.csma.biz/apps/xml_xmldir.php hope that help ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free sitting
In fact, my questions are more about usage than about technical background. For instance, I doubt a user will log his system off when leaving : some don't even turn their PC off. Does anyone has an experience to share about that ? When I tried it, when a user login at a phone, it replaced any previously logged one. hope that help ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best softphone work with Asterisk
I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? You can try my IAX2 softphone for windows : http://www.marccharbonneau.com/asterisk/mediaxphone.php Hope it fits your need ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
Can anyone post a sample of whats needed in iax.conf for an IAX UA to be able to make and receive calls? [7011] type=friend secret=S0m3S3cur3P4ssw0rd qualify=no notransfer=yes [EMAIL PROTECTED] host=dynamic disallow=all allow=ulaw,alaw,gsm context=from-internal callerid=Marc Charbonneau 7011 hth ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. You don't specify if he's on Windows, Linux or OSX. But if he is on Windows, you can try my softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php There is a version using INI file, so you can put all the settings then zip it and send it to him already configured. hth ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
Could I rewrite this in Delphi instead? I never used Delphi to write an AGI but I've seen a class in FreePascal that you could probably use as a base : http://www.automated.it/asterisk/fpc-agi.html hth ___ --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] IP Phone Provisioning Tool by voip.com.sg - xml generation
A. Yes, I have the cojones. He never mentioned what platform it was for. We need something like this for Linux. I got all excited about it only to be terribly disappointed when I unpacked it. From the original announcement : It runs on any modern flavor of Windows. It is not like if he said runs on windows or better, then Linux would seem appropriate ;) You have at least two option beside having a windows machine : 1 - try it in WINE 2 - install Windows in a vmware machine (only way to run windows without ever rebooting your machine) hth ___ --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] SugarCRM, NO!, Foxpro, SI?
First - vtiger is available for those who don't like the SugarCRM licensing. It's not a licensing complaint. At least that has not surfaced. It is more that the programmer does not seem to be comfortable with SugarCRM, MySQL and php. Biggest compliant about sugar is - hard to configure, does not work with latest php. Have a look at vTiger then (fork of SugarCRM). Works with latest PHP and MySQL, easy to configure and is free : http://www.vtiger.com/ hth ___ --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] call dispatching - legacy application
need to preprocess 1) incoming call get caller id lookup some info in my db, 2) based on the result dispatch the call to the right operator step 1 is ok I developped a small .php script that connect manager and parse events, now I have to tell AAH do dispatch call to the right operator From your incoming context, call an AGI and pass it the CallerID. In the AGI, query your DB to find your destination then set an Asterisk variable with the destination. In your dialplan, take that destination and dial it. Something like exten = s,1,Answer ; Answer the line exten = s,n,Wait,1 exten = s,n,AGI(aginame.php,${CALLERID(num)}) exten = s,n,Dial(Local/${MYDESTINATION}) exten = s,n,Hangup() N.B.: Code not tested and written from the top of my head N.B.2: Since you're using AAH, you should put that in extension_custom.conf Also, have a look at the PHPAGI class : http://phpagi.sourceforge.net/ hth ___ --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] Changing Voice from Male to Female
Hi List, I wanted to know if anyone knew of a way with asterisk to switch the voice of a caller from male to female or vice versa. http://www.lobstertech.com/code/voicechanger/ hth ___ --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] Transfer via CTI
Any ideas on this? Closest thing that comes to mind is FOP : http://www.asternic.org/ hth ___ --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] Remastering asterisk
Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Have a look at Kickstart hth ___ --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] Doorphone
Responsibility for answering the door is shared by the entire office. But A) noone wants their phone to ring, there's a door chime) and B) noone specific will accept responsibility for answering the door. So, we need a solution that follow I'm answering the door now, these are the buttons I push. So, when someone is at the door, you call whatever extension to get to the door intercom, talk to them, then you decide to open it. You hangup, then dial an extension that does only this, unlock the door. Something like [door-opener] exten = 555,1,System(script_to_unlock_door.sh) exten = 555,n,Hangup() If you really don't want to have to dial a second extension, look at applicationmap in features.conf http://www.voip-info.org/wiki/view/Asterisk+config+features.conf hth ___ --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] Doorphone
On 3/27/07, Ray Wadkins [EMAIL PROTECTED] wrote: I looked at a call queue, but it didn't seem to work the way I want. Agents need to log into the queue to get calls, seemingly. Of course, I only stopped on the topic for a short period. with the meetme conference, anyone can answer the door from any phone by dialing the conference extension, just not open the door. You can have static agents so they don't have to login, check http://www.voip-info.org/wiki-Asterisk+call+queues Wondering why you don't just dial multiple-phones, like this Dial(SIP/7001SIP/7002SIP/7003) The first one that answer the call is the lucky one. That way, your DTMF signals would work. hth ___ --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] Voicemail mailbox number passed in connection?
Does anyone know how to configure a SIP phone to pass the mailbox number to the voicemail service when dialing? I would like to press the message waiting lamp and be prompted for my password instead of mailbox number. Can this be passed in the set-up call or based on caller-id? based on callerID with something like this : exten = *97,1,Answer exten = *97,n,Wait(1) exten = *97,n,VoicemailMain([EMAIL PROTECTED]) exten = *97,n,Hangup() then just configure your phone to point to *97 (or whatever you choose as this extension) hth ___ --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] play file and action only stop if one defined key has been pressed
I would like that user cann press 3 and then actions can be taken. Problem ist if the pressed key not 3 the user jumps to extension i and then the file will be played from start again. I would like that the play of file is only stopped if the user has pressed the key 3. What for an command can i use to make this happened? check http://www.voip-info.org/wiki-Asterisk+cmd+Background I think the m option is what you are looking for hth ___ --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] Summary of Trixbox vs. custom install
You mean compiling raw tar.gz or SRPMS? And where do you download them from? Trixbox site or the original vendors' sites? I just download the tarball from asterisk.org and compile it. Trixbox is not a special version of Asterisk, it is just an easy way to install Asterisk, FreePBX, FOP and a bunch of other packages. hth ___ --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] Native format prompts
I am trying to implement native format (ulaw) voice prompts and music on hold. Different documentation has different file extensions. Does Asterisk recognise them all? So far I have .ulaw .ul .pcm . Which should I use so Asterisk recognises them as native uLaw files From what I know, .ulaw hth ___ --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] Summary of Trixbox vs. custom install
I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? You can just install CentOS with RAID and whatever you want, then use the Trixbox tar package instead of the ISO. Still, why on earth did the Trixbox team didn't leave the option of doing a custom install with the ISO ? 2) How easy it is to find Trixbox SRPMS? Is it possible to compile new software (i.e. Asterisk or FreePBX, etc..) on Trixbox without having to rewrite all the configuration files, changing all paths, all permissions, and so on... You can update Asterisk/zaptel/whatever by just downloading the source and compiling it. My home system was installed with [EMAIL PROTECTED] on version 0.8 (if memory serves me right) and I upgraded Asterisk and Zaptel to version 1.0.10 by downloading and compiling. I know, this is a really old version and I should upgrade, but hey, it is doing everything I need and it is stable (uptime of 315 days). IMHO, Trixbox can me customized alot, but you need to know where and what to modify. I believe that if you know enough about how Asterisk work, you can get around Trixbox limitations. One thing to remember is that the files you can modify are the _custom.conf files. Never touch the _additional.conf files, they will get overwritten next time you click Apply changes in the GUI. The normal base files (sip.conf. iax.conf, etc) can be modified since the GUI doesn't touch them. But I also think that there is nothing that can beat a plain install as far as customization go. YMMV ___ --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 dual contexts stupidity
Significant albeit insanely stupid Asstricks message: 2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got something to jump out with ('2')! (Oooh how about creating errors we can figure out Digium!) Any thoughts What Error ? it says DEBUG This just tell you that the user pressed '2' Actually, the first time I read that message I was laughing :) Only in opensource product you have the priviledge of having funny message hth ___ --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] Testing IVR / Callcenter applications
We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? Just use an IAX or SIP thrunk to/from another Asterisk. there is no real difference from Asterisk's stand point if the call comes from IAX, SIP or ZAP hth ___ --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] Only secretary can call the boss, all others only reach the secretary when dial the boss extension
How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extension. the keyword is context boss extension : 4321 secretary exten : 4322 in sip.conf for the secretary config, put her phone in the context secretary-context for other callers (PSTN lines, other office exten, etc) put them in context normal-people-context [normal-people-context] exten = 4321,1,Dial(SIP/4322) [secretary-context] exten = 4321,1,Dial(SIP/4321) like this, when someone dials 4321, they will reach his secretary, except when the secretary dials it, she will reach him. This is just an example written from the top of my head on a friday afternoon so it is not tested, etc :) hth ___ --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] dialplan and *
exten = ,n,Queue(|t|||300) exten = *,1,Macro(agent-add,,) exten = **,1,Macro(agent-del,,) So my question is , what means these one/two asteriks (*,** ).Maybe it is like priority.? It means that to login as an agent on the queue you have to dial * and to logout you dial **. hth ___ --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] Wanpipe 2.3.4-2 + kernel 2.6.19 = problems
This is the error i got. I've grepped through all of my include/linux/ wanpipe_includes.h files i have on my server (there is actually a couple of them), and replaced config.h with autoconf.h, but still i get the same error. Looks like I'm unable to locate the include/linux/ wanpipe_includes.h file wanpipe is actually looking for. Is there a patch or a newer version of wanpipe that has this issue solved? From the changelog of 2.3.4-4 released on 2007-01-09 (ftp://ftp.sangoma.com/linux/current_wanpipe/ChangeLog.stable) - Updates for 2.6.18 and 2.6.19 kernels. hth ___ --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] phpagi transfer example
Ok, how can i do the transfer from the caller to $keys ? Probably by using a goto : http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#goto hth ___ --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: FW: [asterisk-users] Get dialed numbers in AGI
All the variables here was my_var, it worked for GET VARIABLE but didn't for SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert my_var value into a db? - What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for digits. - STDIN shoudn't get the result of READ or GET VARIABLE? Where these values go? For AGI in PHP I always use this : http://phpagi.sourceforge.net/ Would make your life easier hth ___ --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] getting tones during conversation
after the Dial has connected, I want the caller (on a SIP phone) to be able to press keys in order to record call status. is this possible? Have a look here : http://www.voip-info.org/wiki/view/Asterisk+config+features.conf applicationmap is what you are looking for hth ___ --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: Is there a low cost cell phone base station for asterisk ?
The phone in question just prepended 010whatever to ALL phone numbers dialled, which makes it pretty crappy to use with a line that does not allow for network selection codes, or on lines that need a 0 for a POTS line. You could use it as an extension on Asterisk and strip that pre-pended number from your dialed number. Nice way to screw their attempt to screw you :) ___ --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] Sangoma Remora A202
Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. Maybe your card is not properly seated. seems to have a lack of documentation, but it may just be me It is just you ;) http://wiki.sangoma.com/ If you still have problems with the card, contact Sangoma, they have very good customer support : http://www.sangoma.com/main/contact hth ___ --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] Sangoma Remora A202
Thanks - that turned out to be the problem. Well- one of those solutions. I removed the blank and swapped the FXO module to the other port. I don't know if it was a bad port on the A200, but since I don't plan on using it, I won't worry about it- just regret it in a year when I get a second FXO module ;) No you won't, since Sangoma cards come with a 5 year warranty ;) Glad you fixed it ___ --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: RE : [asterisk-users] Happy 2007!!!
I wish you all a Happy 2007 filled with an almost-bug-free, full-of-nice-features Asterisk 1.4 :c) ___ --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] Background switch to different context
I am using the Background() function to ask for the extension, but the extensions are in a different context. Is there a way to tell Background() to look for the entered extensions in another context other than the currently running one? in that context you can do include = other-context hth ___ --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: php agi trixbox help
Not sure if this has anything to do with it but running the input.php script directly from the command line gives this warning: PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0 yum install php-imap hth ___ --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] Insert 1+areacode for VOIP calls
Is there a way I can create a _NXX extension and insert 1 and areacode when dialing? exten = _NXX,1,Set(CALLERID(num)=6162997590) exten = _NXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1514${EXTEN}) replace 514 with your area code hth ___ --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] AGI Help Please
Below are a few errors in the script and on a google search, although I found people with the same error, I didn't find a resolution. Any thoughts on what is causing this error? Any thoughts as to why the output is not showing on the CLI without doing a debug? snip Content-type: text/html X-Powered-By: PHP/4.3.9 These 2 lines should not be there. AGI Tx AGI Rx AGI Tx 510 Invalid or unknown command AGI Rx AGI Tx 510 Invalid or unknown command These 2 errors are probably caused by the Content-type and X-Powered-By lines. AGI Rx VERBOSEThere have been AGI Tx 510 Invalid or unknown command AGI Rx VERBOSE125 calls made AGI Tx 510 Invalid or unknown command According to this page http://www.voip-info.org/wiki/view/verbose Usage: Verbose(message [level]) Also, you usually put error_reporting(0); at the top of the script so you won't have warnings and errors confusing Asterisk. I never wrote a PHP AGI without using this : http://phpagi.sourceforge.net/ so I can't help you much You should give it a try, you might like it :) hth ___ --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] Need quality toll free 800 number over IAX?
I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Never used them but the rates seems ok : http://www.les.net/ ___ --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] IBM Server / USB Ports
So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!! I would try moving the Digium card to another slot. Your Ethernet controlled must be onboard and it share its IRQ with the slot where the Digium board is. hth ___ --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] Need help getting started with asterisk
Now for some reason instead of giving me an error on the caller ID, it's not mentioning the caller ID at all. Is there some explicit thing I need to put in to get the caller ID? callerid=asreceived ___ --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] Need help getting started with asterisk
camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoincomingen This should show something like this : panoramix*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en 1from-pstn en so something is missing as Asterisk doesn't see your Zap channel what does your zapata.conf looks like ? ___ --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] AGI and php simple example
I've read http://www.voip-info.org/wiki-Asterisk+AGI+php but i can't understand how to play sounds and read DTMF digits... Have a look at this : http://phpagi.sourceforge.net/ hth ___ --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] Dial 9 For Outside Line?
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1}) exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2}) exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}) Just add a 9 in front, like this : exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1}) exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2}) exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}) hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial 9 For Outside Line?
Just add a 9 in front, like this : exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1}) exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2}) exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}) Oups, pressed Send too fast, here is take 2 exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:2}) exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3}) exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1}) ___ --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] Linux distro + Asterisk or Trixbox?
I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial ease of use isn't really a huge benefit to me. In the end stability and upgradeability will be my main concerns. My favorite for stability and upgradeability is CentOS + Asterisk plain install As a proof, here is what I get on my home PBX [EMAIL PROTECTED] root]# uptime 10:04:01 up 250 days, 19:53, 1 user, load average: 0.00, 0.00, 0.00 I think that speaks for itself :c) I'm not saying Trixbox is not stable (since it is based on CentOS), but it is not as customizable as a plain install (IMHO) YMMV ___ --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] TDM2400
[channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes To the best of my knowledge, all the settings you put after defining the channles (channel= line) are useless. You have to set all the settings BEFORE you define the channels. hth ___ --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] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
I can't risk spending a few thousand just to reach the conclusion that Digium's PRI or BRI cards do not work with a particular system's PCI-X slots/bus... Or, worse, staying with a dead card / system board in my hands ! :-( Anyone ? I don't know about Digium cards, but I just installed a Sangoma A101 card into an IBM server in a PCI-X slot and it is working perfectly. You should ask Digium hth ___ --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] Switching from FreeBSD to Linux - which distro?
Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. CentOS works well for me : http://www.centos.org/ ___ --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] any possibility of Vonage Integration
$25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) You don't have a teenager in your home I guess ;) The teenager girl in my home can easily make more than 3000 minutes of call in a month ! ___ --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] How to stop Asterisk to pick up incoming PSTN signal
Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the functionality to make the call out? [from-pstn] exten = s,1,Hangup ___ --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] How can i processed with Call Snooping,
How can i Processed the call Snooping, it my fifth Requesting and posting to Users, Nobody replies it,,, see http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy hth ___ --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] extension launch into AGI
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when picking up the hand set, the AGI is launched without dialing any digits. check http://www.voip-info.org/wiki-Asterisk+config+zapata.conf keyword is : immediate ___ --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] Live call monitoring
What I'd like to implement, ideally, is that once an incoming call is transferred to a particular operator, the system also calls a manager who can monitor silently. I think you are looking for this : http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy ___ --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] Loosing IAX connection between offices
if its a version 5 or higher, that wont be an option, but if its not, give openwrt or ddwrt a try. Actually, this is no longer true (at least for WRT54G), see http://en.wikipedia.org/wiki/DD-WRT for the official list of supported models ___ --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] 2nd attempt - Return code - How to?
Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not registered from http://voip-info.org/wiki/view/Asterisk+functions : Functions in the below list are marked in red if they are only available in version 1.4 and higher. And STAT is marked in red so I guess you're not running 1.4 ___ --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] AGI PHP Issues (Not new to Asterisk but new to AGI)
I am attempting my first go at a simple AGI application using PHP (Getting Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means a professional standard developer. Can't really say what is wrong with your code since I never did an AGI in PHP without this class : http://phpagi.sourceforge.net/ This should make it more easy hth ___ --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: Rewriting caller ID from database?
I use some custom scripts to do database lookups and rewrite CallerID information. Everything works fine with regard to the CID name, however my Cisco 7960 and Linksys SPA-942 phones do not display the calling number. Instead, they display the called number. This makes the phone's call return feature not work. The calling number and name are both properly displayed on all of the softphone clients that I've tried. Here's the format I'm using to set the CallerID. SET CALLERID JONES DARYL A6508701826 If you're using Asterisk 1.2, see this page : http://www.voip-info.org/wiki/view/Setting+Callerid hth ___ --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 and TDM400P ?
for see if the card answer, what is the process ? since your port is configured to be in the interne context, just add this to this context exten = s,1,Answer exten = s,2,Playback(tt-monkeys) exten = s,3,Hangup watch the console and dial-in. if you get monkeys screaming at you, it worked ! hth ___ --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 incoming call behaviour
I am using asterisk to receive call from a DID provider . In configured everything in freepbx properly and its working . I forwarded incoming calls from did to a certain extension . Now i tried calling from another sip provider to this box , when i call from other provider to my DID number then call reaches asterisk and is sent to configured extension .. however if the extension hangs up without picking then also i am being billed at sip provider ( outgoing one ) . In simple words when people call me then they ( other people ) are billed even if configured extension isnt picked up and hangs the phone. Normally when you call a person and they hang up then you arent charged . Is this asterisk behaviour or is it freepbx dialplan the culprit here ? check your the context into which the calls are coming. if you have an answer line, there is the culprit hth ___ --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] Rewriting caller ID from database?
Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? I made a simple PHP AGI that takes the phone number and query a MySQL table to find the name assigned to this number. I still need to make a web interface to enter/modify the list but phpMyAdmin do the job for now. If you want it, just let me know. ___ --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 as an analog call recording solution (was: Recording outbound analog calls with X100P)
Thank you for the confirmation and the warning about disk space. Now I need to decide between the Sangoma A20202 and the Digium TDM2411. I'm leaning heavily toward the Sangoma card for the following reasons: - It doesn't require a 12V power connector for the operation of FXS modules. Maybe, never used it so I can't confirm, but I doubt it since I'm wondering where it would pull the needed power for the FXS (from the PCI bus ???) - It is compatible with 5v and 3.3v PCI buses. - It shares PCI interrupts properly. - It maintains a single synchronous PCI interface for all FXO/FXS ports (additional daughterboards are added to a backplane bus connector) - It has a better form factor. Depends, if you want to put more than 4 ports in a 1U server, should be easier to put a full-length TDM2400P than try to squeeze 2 A200 cards side-by-side. - It is less expensive. These are rather compelling reasons, but I'd still like some feedback from the list prior to making a purchase. I've seen lots of good comments on the Sangoma cards and on their support. I never used the A200 but I can confirm that Sangoma say they support FAX calls, while Digium say that the TDM2400P was only designed for voice. YMMV Both cards offer hardware echo cancellation for an additional cost. Do the benefits of hardware EC justify the expense in this scenario? Usually when you bridge 2 ZAP channels you don't need echo cancel. I would buy without the echo canceller and if you really have echo that you can't get rid of with software echo can, you could add the hardware echo can after. With Digium's TDM2400P you can add the echo can after. With Sangoma, I think you have to exchange the card for one with it. You should call Sangoma to confirm (http://www.sangoma.com/main/contact) Thank you, You're welcome ___ --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] Recording outbound analog calls with X100P
Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS modules to the existing analog phones. Then with a simple dialplan, all inbound and outbound calls could be recorded by Monitor. I wanted to mock this up using some X100Ps that I had laying around, but found that I could only record inbound calls. I believe that I need an FXS interface to record outbound analog calls but my past experience is with T1 interfaces, so I could be mistaken. Of course you can, if you have 4 FXO and 4 FXS, you could make a really simple dialplan and record the calls that pass through it, incoming or outgoing, and the users wouldn't even know that there is a pbx between them and the PSTN. You will need a lot of space to keep them all, but you could make a simple cron job that would erase any recording older then, say, 2 months. Also, you would have the benefit of having CDR hth ___ --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] Problem with internet down
I believe that the problem really is fault of DNS lookups, but as I should proceed for resolve that?? see the first point at http://www.voip-info.org/wiki/view/Asterisk+administration The best solution for now is probably to have a caching dns server on your Asterisk box or in your LAN ___ --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] Ringing phones
Apologies.. we are using a sangom 4 port FXO card. It used to work (or so the company claims that has the PBX), but they are saying it stopped.. yet nothing has changed on the PBX system. I have verified it IS picking up and then passing the call onto the ringgroup (hence taking it out of the phone companies domain). Matt, check in your incoming context that you don't have an Answer before you dial the ringgroup. If you don't answer and just dial the ringgroup, Asterisk won't pickup the incoming call until a phone in the ringgroup answers it. hth ___ --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] Follow Me problems
Today we appear to have discovered our first bug. We have an extension setup to followme by ringing that extension + an external cell # (ringall). If nobody answers after 20 seconds the destination if no answer is set to go to the extensions voicemail in the followme module. The problem is it just keeps ringing forever. If we delete the followme it forwards to the voicemail as per the default SIP extension configuration with voicemail enabled. Anyone run into this? Is there a workaround? Any advice would be greatly appreciated as always. Our configuration is: Supermicro Pentium D 2.66 Server with 2x512MB Memory 3ware 8006-2LP Hardware RAID 1 Sangoma A200D with 8fxo (latest firmware/drivers as of last week) CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 FreePBX 2.1.3 When Asterisk dial the Cell phone, it goes out on the ZAP channel (Sangoma A200D), so as soon as it hit that channel, the call is considered answered even if the cell phone never actually pickup the call. I didn't play with the followme module myself but that is what I suspect is happening. Just watch the console and you should see something like Zap/1-1 answered ... hth ___ --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 web interface is not parsing the PHP pages
After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . The problem is probably that you didn't install PHP yum install php ___ --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] Compatability
I have a new client who has an existing Asterisk PABX and is looking for us to install a TE110P for him, However he has a Dell SC420 and I have never used one before. I have had no problems with any other Dell servers which we use almost exclusively. Has anyone had any good/bad experiences with the SC420 in relation with Digium cards? According to Digium this model is partially incompatible : http://www.digium.com/en/docs/misc/compatibility_notes.php I would suggest a Sangoma card if you want to avoid problem with that server http://www.sangoma.com/datasheets/p_aft-et1-specs hth ___ --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] No ring tone when using IAX
Then what would be a better solution? Usually the IAX phone will play you a ring tone until the other end answer. If you're phone doesn't do it, then it is a flaw in that phone. What phone is this ? ___ --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] operator console
...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? Have a look at FOP : http://www.asternic.org/ ___ --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] Wildcard X100P Suport
Is the Wildcard X100P still supported? I have one sitting around that I bought 3+ years ago and never used it. I need the functionality now. Before I run off and buy something new, I'm curious if this will just work. It still works with the latest Zaptel (1.2.10) I also have an old TDM400P with 2 FXS modules that I bought at the same time. Then, there was no FXO module for the 400P. Will a TDM400P this old support a new X100M? I don't think that something changed, so it should work. You should contact Digium to be 100% shure hth ___ --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] voicemail usernames can't begin with j letter?
Thanks to all that replayed, I made like Mr Watkins told me, and my problem is apparently solved, although, because of the usage of the syntax VoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and vm-isunavail, while before were only played vm-intro and beep. Is there a way to disable this two other files that get played every time? see http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail ___ --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] Getting started with sample dial plans
Now I'm ready to begin playing with dial plans and am having a difficult time getting started. You may want to read the book : http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 That should help you ___ --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] Why is this happening?
You've got a very poor grasp on how things work. Please don't pretend to know what you're talking about. # netstat -apn | grep :80 tcp0 0 0.0.0.0:80 0.0.0.0:* LISTEN 782/httpd tcp0 0 204.xxx.yyy.188:8080.xxx.yyy.167:58620 ESTABLISHED 814/httpd tcp0 0 204.xxx.yyy.188:8062.xxx.yyy.15:55384 ESTABLISHED 1068/httpd tcp0 0 204.xxx.yyy.188:80165.xxx.yyy.230:4392 ESTABLISHED 1084/httpd tcp0 0 204.xxx.yyy.188:8065.xxx.yyy.111:6982 TIME_WAIT - tcp0 0 204.xxx.yyy.188:80200.xxx.yyy.43:8198 ESTABLISHED 817/httpd tcp0 0 204.xxx.yyy.188:80165.xxx.yyy.230:4304 ESTABLISHED 815/httpd As you can see, I am *still* listening on port 80 and have numerous connections from different systems, even numerous connections from the same system. I am really sorry, I've read that explanation somewhere and it made sense. Now that I've been corrected, I won't make that same mistake again. Please excuse me. The one that never did a mistake, never did anything ___ --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] Remote UNIX connection, Remote UNIX disconnected displayed every second
Hi, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? You probably have some script that use the console to query something, like the WebMeetme application. hth ___ --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] Why is this happening?
Why is it running on port 1207? because Asterisk is listening on port 4569 and when a connection comes in, it as handed to another port so it can continue listening on port 4569. Otherwise you would only be handling 1 connection at a time. Pretty basic networking stuff I think :c) ___ --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] Why is this happening?
Thanks for the answer, but I don't buy it. There are currently 0 calls up on that bridge, while another connection which has calls up on it is on Port 4569.. please try again. IAX2 is suppose to run on ONLY one port.. this is why it is so nice for use in firewall situations. It doesn't change a thing ! Same thing happens with a webserver. It listen for connections on port 80 (default port) and when a connection comes in, it is handed to another free port on the server so the main server can continue listening on port 80. Same thing with FTP, etc. All TCP servers that accept more than one connection I think that what iax2 show peers display is the remote port from which the client connected. iaxclient library defaults to using port 4569 as the originating port but there is a function to specify another port. Check on your machine while you're surfing the web, your browser doesn't use port 80 as the originating port. Connect to an FTP server and check your netstats, you'll see that you're not connected to port 21 on the remote server ___ --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: Re[2]: [asterisk-users] Why is this happening?
On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote: OMG, please read more about network ports. Could you tell me what is wrong with my explanation ? ___ --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] OT: Hand free solution recommandation
Ideal would be a headset audio+microphone with RJ11 4p female that we could plug into the handset cable of any IP phone, or a converter 2xjack2,5mm female RJ11 4p female -which seems not to exist-. What are you recommanding/using/installing in such case? I don't know if it would work on any phone, but it works on Cisco 7940/7960 : http://www.mml.uni-hannover.de/einhorn/headset/index_e.html hth ___ --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] MODEM (data) througt asterisk ?
Is it possible to connect a modem to a remote service through asterisk ? Basicly to ilustrate : Accounting department need to connect with analog modem to their bank to order some wire transfert. Modem - Chanel Bank FXS - Asterisk - TDM2400 FXO - Modem in remote site. If you get it working you're lucky. Digium's official statement on the TDM2400 is that card as been designed for voice calls, we don't support data calls You would have better luck with a Sangoma A200 hth ___ --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] pop a web page with DID in url
I'm looking to do this. When a call comes in to an agent in a queue, pop a web page like this http://www.mydomain.com/cgi-bin/script.cgi?did=952900 Where did is the number the caller dialed to reach the system in the first place. I know Hudlite can do this we caller ID, but the DID feature is not there yet. Does anyone have any other software they know of that can do this? Some softphones support handling URL when you pickup the call. You can set that URL to anything you want from the dialplan. shameless-plug My MediaX softphone (current beta version) support it. Let me know if you want to try it /shameless-plug hth ___ --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] Call Interception
I'm deploying an asterisk PBX for a Call Center and i was ordered to check if the Customer Support Supervisor could intercept the calls so they can check how they employees work with Asterisk. have a look at these : http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge and http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy hth ___ --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] Dialplan Syslog
It'd be cool if someone wrote a syslog() dialplan application for Asterisk *hint* *hint* That could be usefull, but what is wrong with : System(logger Asterisk can use syslog) ? ___ --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] Right way to prevent analog channel from answering the phone?
Is there a more elegant way to tell it to answer/not answer on command? Put your Zap line in a context that do just this : s,1,Hangup() hth ___ --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] IAX phones?
Just wondering if there are any IAX phones worthy of the name phone out there -- looking for hard phones, but I suppose a Linux-based softphone wouldn't, you know, hurt. ;-) Idefisk looks pretty nice and there is a Linux version : http://www.asteriskguru.com/idefisk/ There is also iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/index.html Also, check on iaxclient page : http://iaxclient.sourceforge.net/ hth ___ --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] Variable that gives the SIP channel
What I would like to do is in my flash hook dialplan code to ass something like Hangup(SIP/100-fe65), but where can I get that SIP/100-fe65 ? Is there a variable set with this information available in the dialplan ? ${CHANNEL} have a look here : http://www.voip-info.org/wiki-Asterisk+variables hth ___ --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 7940 Problem (Mess)
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll work on my Asterisk box (outside of the FXO FXS modules on the TDM card in the Asterisk server, I only run SIP on the hardphones). I don't know the phone's password (sound familiar?). - Have tried everything, cisco, *##, etc Nothing works. You could factory-reset the phone. try this http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml and this http://www.sokol-associates.com/?q=node/51 hth ___ --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] Modem calls
I need to pass modem calls through a TDM400 card. Conecting the modem to the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4) directly. According to Digium, Fax calls (and modem calls) are not supported on the TDM400 or TDM2400. They are designed for voice only. If you get it to work, you're lucky. Sangoma test all their cards with faxes, so maybe you should try their card. For your problem, run zttest and adjust everything to try to obtain 100%, this may make it work. hth ___ --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] sound file length
At some point in my dial plan, I need to find out the length of a sound file in seconds (to weed out things that are way too short) the record application doesn't seem to have any facilities to do that. any ideas ? use sox beep.wav -e stat and parse the output man is your friend google also :) ___ --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] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
We have a problem in configuring Sangoma A104. We want the 2 ports to be configured as E1 and the 2 ports as T1. If I'm not mistaken, you can't do that with the A104D, that's why they sold me 2 x A102 for the same price as a A104. Better check with Sangoma. hth ___ --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] detecting a users number using the dialplan or AGI
keeping track of the confno is easy since I created it, but I don't know how to determine the user number of the last person that joined the conference. Is there a way to store this in a variable before they join the conference? Or perhaps a way to detect the last user to join the conferences number? Maybe by listing the users in the conference and parsing the output something like : meetme list 87004 you will get an output like : User #: 1 Channel: SIP/7004-1d3f (Admin) hope this help ___ --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] 7970 'LoadID incorrect' problem
Does ANYONE have any clues? Only played with 7940 and 7960, but I will try to help since nobody comes forward loadInformationSIP70.8-0-3S/loadInformation Shouldn't that be something like P0S3-08-2-00 ? ___ --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] Monitoring/Listening In
I wish to setup asterisk for training purposes so that I am able to listen in to an extension while a call is going on? http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge and http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy ___ --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] About IVR and Oracle
On 8/23/06, Infobox Peru [EMAIL PROTECTED] wrote: maybe you could make it with PHP and its driver for Oracle. For PHP have a look here : http://phpagi.sourceforge.net/ ___ --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] Text to Speech
N.B.: Please use plain text when sending to this list Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better then Festival. You have flite that is free and, IMHO better than festival (http://nerdvittles.com/index.php?p=134). I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting for the Linux version as I don't have time to babysit a Windows server :) hth ___ --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] Text to Speech
All I can find for Flite is for AAH, does it work as well with plain Asterisk? Is the setup the same? Never tried it, but it should be the same. Have a look here : http://dialogpalette.sourceforge.net/extras.html hth ___ --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] astbill white screen!!
I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. you have to enable it in php settings. Go in /etc/php.ini - change setting error_reporting to E_ALL - change setting display_errors to On - restart apache now, at least, it will tell you what goes wrong N.B.: display_errors should not be enabled on a production server hth ___ --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] Rookie question, trying to learn
The problem a number of people are not entering the pin fast enough ,they are not given enough time to enter the PIN( I assume this is a mailbox number) looking at all the doc is seems everything is configurable, can some one point me in the right direction of where to start looking? check http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout and http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ResponseTimeout Hope this help ___ --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] Examples of handeling input from phones with PHP
Can anyone direct me to where I might find examples of handling interactive input from a phone using PHP and AGI. I want to have someone dial an extension and then have the system request input from the user, take that input and put it into a database. Start here : http://phpagi.sourceforge.net/ ___ --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] phpagi problem
#!/usr/bin/php -q ?php require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi-say_digits(62410); $cid = $agi-get_variable(dir); $agi-say_digits($cid); ? I'm getting this error: parse error, unexpected '=' on line 6 I don't know why you're getting this error, it parse correctly here. But one thing is that the line $agi-say_digits($cid); won't work. When you do a get_variable, the result you get is an array, and the member holding the value is 'data'. So you have to write your line like this : $agi-say_digits($cid['data']); Check the documentation : http://phpagi.sourceforge.net/phpagi2/docs/ hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can one SIP extension be used for two phones?
Picture this: Exten = 100 #My Phone Exten = 200 #MythPhone Call comes in. Dialplan calls both extensions. MythPhone is an add-on for MythTV,so when i receive a call,the CallerID is flashed up on my TV. I want to add another MythPhone to my other MythTV box upstairs. Do i have to make a third extension and get the dialplan to call all three extensions or can the second MythPhone instance also log into Asterisk using the same extension? Each phone needs its extension But, you can have the dialplan ring more than one phone and the first to pickup the call is the lucky one :) like : Dial(SIP/100SIP/200SIP/300) hth ___ --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] Global variables and AGI
Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Reading this : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set you need to use the g option If you can't use it from phpagi, you could, at worst, set a local variable from your AGI, then in the dialplan take the value of that one and apply it to the global one using option g hth ___ --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