Re: [asterisk-users] Asterisk Redundancy
I’ve been googling “asterisk redundancy” but all I’ve found is questions, and no real answers. Is this any help Dan? http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions Chris -- _ -- 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] Siemens Gigaset + Asterisk Query?
Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to their Asterisk SIP subscriptions? Hi Al, I've played with the Siemens Gigaset in the past and I don't recall being able to do this. Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP password encryption
A really, really quick question here! Am I right in thinking that all passwords sent across the network in Asterisk are MD5 encrypted without me having to specifically set anything up to make it happen? Thanks so much! Chris ___ -- 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] SIP password encryption
A really, really quick question here! Am I right in thinking that all passwords sent across the network in Asterisk are MD5 encrypted without me having to specifically set anything up to make it happen? Thanks so much! Chris Sorry, I forgot to add when using the SIP protocol ___ -- 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] SIP password encryption
On Mon, Feb 9, 2009 at 9:28 PM, Kevin P. Fleming kpflem...@digium.comwrote: Chris Rowson wrote: Am I right in thinking that all passwords sent across the network in Asterisk are MD5 encrypted without me having to specifically set anything up to make it happen? The simple answer is 'yes', the correct answer is 'no' :-) MD5 is not encryption, it is a digest (hash) function. What happens in SIP (and HTTP basic auth) is that the shared secret (the password) is run through a supposedly secure digest function (MD5), along with a shared non-secret value (the nonce). The result of this digest function is then sent to the other party, which does the same calculation and compares the result. If the result matches, then the shared secret must have been the same. So, since your goal is to avoid the secret being sent unprotected, that is the case; the password is *never* sent across the wire, even when encryption is in use (SIP over TLS, for example). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org Thank for taking the time to write such a comprehensive answer Kevin! Cheers Chris ___ -- 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] Looking for SIP loud ringer
On Wed, Jan 28, 2009 at 3:31 PM, Steve Gladden aster...@michiganbroadband.com wrote: If you wanna go low tech. down dirty you could also go with a conventional POTS phone line 'loud ringer' device and simply hook it to an ata such as a PAP2, and add the PAP2 to the ring group. I'd go for an ATA too ;-) Chris ___ -- 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] CentOS and BAT File
In windows, we use BAT file to execute few series of command , which help us in not writing each command manually everytime we want to execute those commands. In CentOS, I want to do the same thing. Any Advice ? ___ Start here: http://tldp.org/LDP/Bash-Beginners-Guide/html/ It's called bash scripting. You can create a file which contains a list of commands that you want the system to perform. You can even use a system called Cron to have the system execute your bash script at a specific time. http://www.linuxhelp.net/guides/cron/ To be fair, this is likely a little out of scope for an Asterisk discussion list, but you might get more help over at the Centos website http://www.centos.org/ Have fun! Chris ___ -- 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] CentOS and BAT File
*1) What name I have to save it.Like what extension ?* extension? that isnt important but the common is to use .sh for shell scripts That's one of the strange things you'll notice if you're used to a Windows environment. Under Linux, it doesn't matter if you give your script a file extension or not. Linux still knows what to do with it. ** 3) How I save it ? open any text editor write and save *2) How to run it to execute it ?* set runing permisions chmod +x . and then ./name there is a dot before the slash. you should read the links about bash programing, it is a very usefull skill for linux admin. and probably you MUST have that skill ___ -- 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] Call Recording - Asterisk
I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Chris, Make sure that all of your SIP clients are set to canreinvite=no in sip.conf. The default is canreinvite=yes, which allows RTP to bypass Asterisk. Certain things (codec translation, playback of audio files, etc.) require Asterisk to be in the RTP path, which may explain why you're recording some of the calls. If you're still missing calls, make sure Oreka is configured properly in config.xml. In particular, the AllowedIpRanges and BlockedIpRanges settings provide IP address filtering at the Oreka level. In general, I've had to configure these to prevent getting two recordings of each call (since Asterisk acts as a B2BUA) but your configuration may be too strict. Running tcpdump/Wireshark on the Oreka server will let you see exactly what's being mirrored. There is even a setting in Oreka named PcapFile that will let you playback the packet capture file over and over until you're satisfied with your configuration. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matthew, Thank you so much for your advice. It's really appreciated - I'll go through it and see where I get. Thanks again Chris ___ -- 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] Call Recording - Asterisk
Hello folks, I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Just trying to get this sussed out in my head! Thanks for your time. Chris Hi again, didn't get a reply to this one. I'm a bit stumped so I thought I'd try the list one more time to see if anyone has an answer. If not, thanks for reading anyway! Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Recording - Asterisk
Hello folks, I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Just trying to get this sussed out in my head! Thanks for your time. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk voicemail
Hi folks, I'm working on a solution using the Asterisk voicemail component and wondered if anyone knew the answer to this question please? I understand that Asterisk saves voicemail to /var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I wondered if * creates the file in memory (or tmp/or wherever) and then loads the completed file into that directory, or if it writes the file to the directory directly, appending it till the recording is finished? Hope that makes sense! Cheers Chris ___ -- 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] Asterisk voicemail
Hi folks, I'm working on a solution using the Asterisk voicemail component and wondered if anyone knew the answer to this question please? I understand that Asterisk saves voicemail to /var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I wondered if * creates the file in memory (or tmp/or wherever) and then loads the completed file into that directory, or if it writes the file to the directory directly, appending it till the recording is finished? Sorry to reply to my own post! I notice a tmp directory at /var/spool/asterisk/voicemail/context/mailbox/tmp/ I'm wondering if this is where the file is created, and then moved to the INBOX folder perhaps? Chris ___ -- 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] Asterisk voicemail
I'm working on a solution using the Asterisk voicemail component and wondered if anyone knew the answer to this question please? I understand that Asterisk saves voicemail to /var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I wondered if * creates the file in memory (or tmp/or wherever) and then loads the completed file into that directory, or if it writes the file to the directory directly, appending it till the recording is finished? Sorry to reply to my own post! I notice a tmp directory at /var/spool/asterisk/voicemail/context/mailbox/tmp/ I'm wondering if this is where the file is created, and then moved to the INBOX folder perhaps? That is exactly how it works. It is created with a random file name that is guaranteed by the API to be unique, the messages are recorded there, then when complete, the files are moved to the INBOX directory. This was initially done to avoid the problem of mailbox owners checking voicemail while a voicemail was being left. In certain cases, it would cause possible collisions when a mailbox owner deleted a message while it was being recorded, which left orphan (and incomplete) .txt files in the INBOX directory (since .txt files are updated after the message is complete). -- Tilghman Thanks Tilghman, that's really helpful :-) Chris ___ -- 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] Asterisk voicemail
Hi folks, I'm working on a solution using the Asterisk voicemail component and wondered if anyone knew the answer to this question please? I understand that Asterisk saves voicemail to /var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I wondered if * creates the file in memory (or tmp/or wherever) and then loads the completed file into that directory, or if it writes the file to the directory directly, appending it till the recording is finished? Sorry to reply to my own post! I notice a tmp directory at /var/spool/asterisk/voicemail/context/mailbox/tmp/ I'm wondering if this is where the file is created, and then moved to the INBOX folder perhaps? Just do an `asterisk -r` and leave a voicemail. It answers all your questions. Hi thanks. I did think of that, but currently I don't have access to an asterisk box to test on. Just sent this message on the off-chance that someone might happen to know the answer off the top of their head so I could get back to someone else! I'll just try it later when I do have access. Chris ___ -- 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] Asterisk voicemail
Hi folks, I'm working on a solution using the Asterisk voicemail component and wondered if anyone knew the answer to this question please? I understand that Asterisk saves voicemail to /var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I wondered if * creates the file in memory (or tmp/or wherever) and then loads the completed file into that directory, or if it writes the file to the directory directly, appending it till the recording is finished? Sorry to reply to my own post! I notice a tmp directory at /var/spool/asterisk/voicemail/context/mailbox/tmp/ I'm wondering if this is where the file is created, and then moved to the INBOX folder perhaps? I'm not sure about voicemails, perhaps they have temporary storage in /tmp/, however there's more general option for asterisk. See man asterisk, there's command -t which could be passed at asterisk startup, then asterisk will write all files in /var/spool/asterisk/tmp (allocating empty filename before), and after recording finishes it will move them to correct location. Regards, Atis Thanks Atis, I just brought up the Asterisk man page online and see the command you're talking about. -t When recording files, write them first into a temporary holding directory, then move them into the final location when done. I'll give it a try and see what it does. Chris ___ -- 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] Asterisk For Windows ?
realy you are helping me! thanks for your help! i try this fitur and i respond to you after trying it thanks! - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 12, 2008 10:11 AM Subject: Re: [asterisk-users] Asterisk For Windows ? On Sun, Oct 12, 2008 at 09:44:30AM +0100, Meftah Tayeb wrote: hi my friend, thank you for this id now i have a screen reader named orca working with all OS tha have GNOME Desktop i decided to work with debian or UBUNTU please, provide to me a best / reliable / easy to use linux distribution i have only one problem: if i try to install ubuntu with ORCA screen reader, orca dont speak the installation step please, try to provide to me a Linux distribution tha have speakable setup user interface A keyword for your searches: brltty (a short for Brail TTY, which is a sort of a hsorthand for a terminal device). For instance the following message claims that the Debian installer should support it, as long as you write from the boot prompt: install brltty http://speech.braille.uwo.ca/pipermail/speakup/2007-May/042945.html Also, if you are using Ubuntu and get stuck with anything, you can use the Ubuntu 'Answers' helpdesk where a community member will try and guide you through any problems you may have. https://answers.launchpad.net/ubuntu/ Cheers Chris ___ -- 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] Asterisk For Windows ?
hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) Why use Windows? If you want something free and easy to use, download a pre-built Asterisk Linux CD. You could try download a Trixbox .iso and give it a go. I'm sure they're are others too... Chris ___ -- 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] 2000+ user Asterisk PBX
Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs Femi I would be interested in some of the replies if you wanted to continue the topic on-list... Your problem might help someone else down the line. Me too, Any reason you want this off the list particularly? Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming call does not reach asterisk.
Hi, this is my first post to the list, but I have tried to search elsewhere for a solution SNIP I'm using sipgate.co.uk for incoming calls, but when I make a test call from the PSTN, the call just dies without connecting to my Astlinux box. (I'm monitoring asterisk console via 'asterisk -rv' and see nothing). SNIP Thanks for the suggestions. I ran tcpdump and it indicated that traffic on that port was being forwarded to the asterisk server. It looks like I basically wrote a load of nonsense in the extensions.conf file. I edited the file to input the extension the incoming call should be coming from and it now works. Working file --- [from-pots] exten = 277,1,Answer() exten = 277,n,Wait(3) exten = 277,n,Playback(tt-weasels) exten = 277,n,Hangup() So in summary it was basically me misconfiguring the box... Cheers Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming call does not reach asterisk.
Hi, this is my first post to the list, but I have tried to search elsewhere for a solution, and have had a read of 'Asterisk - The Future of Telephony'. So you could say that I have at least tried to RTFM as it were! I've configured a couple of Asterisk instances on both Debian and CentOS based VPS's, and got them working fine. However, I recently installed a copy of Astlinux and installed on a WRAP board and I'm totally stuck! I'm using sipgate.co.uk for incoming calls, but when I make a test call from the PSTN, the call just dies without connecting to my Astlinux box. (I'm monitoring asterisk console via 'asterisk -rv' and see nothing). I wondered if it might be a problem with Asterisk not listening properly, or perhaps a problem with my home firewall. Would anyone be kind enough to advise me as to where I may have gone wrong? Thanks, Chris. My sip.conf looks like this: -- [general] context = default ;default context for incoming calls bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes disallow=all;disallow all codecs allow=alaw ;except alaw (1st pref) allow=ulaw ;and ulaw (second pref) register = 277:[EMAIL PROTECTED]/277 [sipgate] ;sipgate sip in on 01482 77 type=peer context=from-pots fromuser=277 username=277 authuser=277 secret=*** host=sipgate.co.uk fromdomain=sipgate.co.uk dtmfmode=inband insecure=very canreinvite=no disallow=all allow=alaw allow=ulaw nat=yes qualify=yes - My extensions.conf looks like this: - [general] static=yes writeprotect=np autofallthrough=yes clearglobalvars=no priorityjumping=no [from-pots] exten = s,1,Answer() exten = s,n,Wait(3) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() -- and netstat looks like this -- Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 *:www *:* LISTEN tcp0 0 *:ftp *:* LISTEN tcp0 0 *:ssh *:* LISTEN tcp0 0 *:https *:* LISTEN udp0 0 *:1025 *:* udp0 0 *:1026 *:* udp0 0 *:1027 *:* udp0 0 *:1028 *:* udp0 0 *:1029 *:* udp0 0 *:1030 *:* udp0 0 *:1031 *:* udp0 0 *:1032 *:* udp0 0 *:2727 *:* udp0 0 *:4520 *:* udp0 0 *:5060 *:* udp0 0 *:tftp *:* udp0 0 *:4569 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:ntp *:* - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
