[asterisk-users] Programming soft buttons on the IP601?
When the IP601 is sitting unused, it uses the first 2 of the 4 soft buttons under the screen. The third one is empty, which is good because it is used for Exit. I would like to be able to use that 4th button for group pickup (*8#) and have it read Pickup. Is this possible? If so, how? Thanks, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax pdf attachment naming badly when no caller id on zaptel
I just installed a Digium TDM04B card. The lines that are plugged into it do not send caller id. As a result (i believe) of no caller ID, the name of the fax pdf is .pdf. The subject is Fax from. Windows clients do not like opening files with just an extension. Is there a simple way to change the name if it comes in blank? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good phones for outside of the office?
I am looking for phones that work well (or at all) when outside of the network and behind a router, such as at someone's home or in a hotel. My Polycom IP601s do not seem to be up to the task, so I am hoping that there is a good alternative for my outside sales people to use to talk to my asterisk server. Thanks in advance, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Taking a Polycom IP601 home
I am taking a Polycom IP601 home to try to figure out how to provision it outside of the office for our outsides sales people. Our asterisk server has a direct outside IP. The IP601 will be behind a router at home so it will not have an outside IP. I am fully opening the company firewall for my home IP so that all services (ir the FTP server) will be available to the phone. What else will I need to have in place to get the phone to work? Has anyone done this yet? Can it be done? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemailmail hanging after entering password
I had a problem with the voicemail system hanging after certain users would enter their password. I found that lock files get left behind. In order to fix this, in my startup script I put this line: rm -f /var/spool/asterisk/voicemail/*/*/*/.lock* Works nicely. Hope it helps someone else. W ___ --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] Dell hardware ...
Alan Bunch wrote: I was going to use a Dell 1425 for Asterisk build but I see on Digium's website that hardware may be problematic. Can anyone shed a litle more light on the problem. I see the Intel ethernet cards seem to cause problems. If I need to disable the onboard Intel on the Dell hardware I can I just need to know what to expect. How about the 850, any word there ? I am running trixbox 1.1.1 on an 850. Actually on 2 850s, one is a hot-swap spare. No problems. The problems that were reported earlier on were with specific Digium boards on specific Dell machines. I have a T-1 for my connectivity which comes in through a Cisco router and have no problems. Look here for more info: http://www.digium.com/en/docs/misc/compatibility_notes.php W ___ --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] Trunk with multiple IPs?
In my case it was not a class c, but just 4 separate addresses, one each in NY, Seattle, Miami and London on the Level 3 network. I ended up creating separate entries for each, in and out, and for the outbound route, put all 4 in the order of their ping times. That is working nicely. W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trunk with multiple IPs?
How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variable to show caller id for a current call?
Is there a variable that can be gotten with GetVar to show the callerid of the current incoming call in progress at a sip extension? For instance, a caller from 516-922-9463 calls extension 234. I would like to be able to be able to get back the 516-922-9463 if I pass 234. Also, can this be done while the extension is ringing? TIA, Warren ___ --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 to show caller id for a current call?
But how do you get that with GetVar? I am trying to do this through the
API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a program
fired off when an extension rings that will have the caller id passed to
it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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[asterisk-users] Speed dials on Polycom IP601?
I just got my first IP601 and put together my first * system (yay!) I have the first 2 buttons set up to be for the extension for the phone. I was wondering how I could make the remaining 4 into speed dials? IE: label button 3 Sales mgr and have it dial extension 246. TIA, Warren ___ --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] zaptel on dual processor, How?
Matt Riddell (NZ) wrote: Warren (mailing lists) wrote: The proper method is, as root, type: yum install kernel-devel The problem is, the kernel headers will have the name 2.6.13-15.8 whereas uname -a will report 2.6.13-15.8-smp. You may need to create a symbolic link. Sorry... yum install kernel-smp-devel You will then have a directory of /usr/src/kernels/2.6.13-15.8-smp-i686 (or somthing like that). You might want to create a link to /usr/src/linux from there to keep things simple. W ___ --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 1.2.10 and Zaptel 1.2.7 released!
Last week I had asked about which * version to use. The response was that if using queues, 1.2.4 was stable and another response stated that 1.2.9 was stable with queues as long as CallBackLogin was not used. Has this been addressed in 1.2.10? Is it even accurate or should I be looking to deploy 1.2.4 for stability? W ___ --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] zaptel on dual processor, How?
Olivier Picquenot wrote: Zeeshan Zakaria a écrit : It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686 Then you might want to use yum to install the apropriate package, the one that contains the kernel source, or at the very least the kernel headers . Or you might grab it on a Cent OS mirror, for exemple: ftp://ftp.dedibox.fr/centos/4.3/updates/i386/RPMS/kernel-devel-2.6.9-34.0.1.EL.i686.rpm I'm no Cent OS expert, but that should be the right rpm . The proper method is, as root, type: yum install kernel-devel Regards, Warren ___ --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 do you harden an Asterisk install?
Sorry - I misread it. Have you ever had a network card fail in a way that did not lock up every network-bound job on the system? I would think that it would be unlikely that you could recover from that easily. Yes, redundant drives with RAID-1 is good. If those drives are hot-swappable and the RAID is hardware-based, it is even better. Having a dual-processor setup is also good - I know many servers that have had a CPU die and still worked OK with only 1. Redundant power supplies are good, but only if they are hooked up to separate UPS's. W shadowym wrote: Thanks for the suggestions but I specifically asked for options OTHER than a second server. Your suggestions about disabling un-needed services are good though. I already do that. I am hoping someone has some suggestions that are not as obvious that I have perhaps not thought of. ___ --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 incoming alternate rings be discriminated?
FYI: If you get no help here and want to search, in the US this is called Distinctive Ring. W Paul Lakra wrote: In the UK I have 'Call sign' from BT on my house POTS line. This gives two different nos. to call the same physical line. One no. gives the usual UK ring-ring, ring-ring, etc. pattern. The other no. gives a ring,ring,ring north American style of ring. I would like to be able to detect the ring pattern and route the calls differently after Asterisk answers them. Just for reference, Call Sign is often used for a separate no. for a fax machine and hardware boxes are available for about £40 (when I last looked - may be cheaper now) to split the line into 2. I would also need an additional FXO/FSO port. Would be nice to do it natively in Asterisk. ___ --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 do you harden an Asterisk install?
Rich Adamson wrote: [-snip-] Then, back up your config files on something else and wait for your server to be compromised. ;) For cases where you expect something to be compromised, and potentially overwritten, perhaps by an automated script, a trick that I have found worthy of using is to move all of the writable files to somewhere (should be /var) and put all of the read-only files under a single directory structure. Then take that structure and make an iso file system out of it with mkisofs. Now remove that filesystem and just leave an empty copy of the root directory. From then on, mount the iso file read only onto the root directory using the loopback device onto the directory in question on boot before the service starts to run. W ___ --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 + mysql 5
The Centos-plus repository has mysql5 rpms. W varun wrote: Hello, Our asterisk server is on Centos 4.2 We want to use Astbill. Astbill requires Drupal and mysql 5. I could not find rpms mysql5 for centos. We are getting mysql extensions issues because of php-mysql. How do we solve this ? Any other billing software that similar to Astbill ? Thanks Varun ___ --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 ___ --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 incoming alternate rings be discriminated?
Paul Lakra wrote: Thanks Warren - I think I found the answer here: http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels It is part of Zaptel configuration. Glad to help. Why don't you update the wiki now and add dsomething about the different name and where it is called that? This will help future users searching as you initially did. Regards, Warren ___ --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] New VEGASTREAM 400 Devices Cheap
Michael Workman wrote: The Vega 400 connects [-snip-] Did you not catch the name of the list? What part of Non-Commercial Discussion did you not get? Post this over on the -biz list where it belongs. W ___ --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] NuFone, please send the log file
Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work, etc. off of this list? This is not the place to discuss your experiences with _any_ company, it's a place to talk about Asterisk and using Asterisk. Please move flamewars and similar discussions to some other forum. I agree with you! Which place is in your opinion the right place? As long there is no other place, such messages will always pop up. How about the Asterisk-biz list? W ___ --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] New York city Asterisk consultants
Dean Collins wrote: If there are any freelance asterisk consultants (or small companies) who Does anyone pay attention to the non-commercial part of the list name any longer?!? HELLO PEOPLE - THERE IS A -biz LIST FOR STUFF LIKE THIS!! W ___ --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 version: 1.2.9.1 or older?
So let's cut to the chase here... If you want to run a production server with queues, which version should you be running to get 30+ days of uptime without needed a reset? W ___ --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 do you harden an Asterisk install?
shadowym wrote: I remember reading a small write up somewhere. I think it was on the Asterisk Wiki. I can't find it anymore. It's probably a bit dated by now but some of it would still be relevant. Can anyone recommend a good guide or even some of their own suggestions. For clarity, what I mean by hardening is to make an Asterisk Server or network appliance or embedded server or whatever you want to call it, as fail safe, stable, and reliable as possible. Just like an expensive traditional PBX. This is for a small business application of 50 extensions or less. It can't be too crazy like redundant servers or anything like that. I am looking for ideas like RAID 1, redundant power supply, cron job to reboot every night (yuck!), disable caching(?), Astlinux on embedded with CF, yada yada! Anyway to set up automatic failover to a second Network Card with same IP if primary network card fails? That is one point of failure I haven't found a way around yet. Failure of the managed switch is another one I get a bit paranoid about. Switches generally don't fail but I'd like to have some sort of fail safe plan. ___ --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 You are talking about 2 things: (1) How to harden a linux box (2) How to do failover. for (1), be sure telnet, ftp and any other service you do not need is off. Move standard services to non-standard ports, especially web and ssh. Do not run a name server on the box. For (2): You need to have a secondary box that runs a mirror copy of Asterisk and mysql and pretty much has everything else configured the same. mysql should be replicated to the second box. You then run a program on the second box that pings the first box. If the first box fails the second takes over the first box's IP and runs with it. There are heartbeat programs that can help out with this. W ___ --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] comcast info -- somewhat offtopic
Derek Whitten wrote: i just upgraded to 8M and my avg d/l speed went up to between 850KB/s - 1.05MB/s That sounds about right. If you are download a 1MB (MegaByte) per second then you are at 8mbits per second. LIne speed is always in bits (little b). I use Optimum Boost (Cablevision's 2/30 service) and truly do get my 2 upstream. I don't have anywhere to test 30mbit download with, but I really do not need that - I just needed the 2 mbit up to support my bittorrent (legal) habit. W ___ --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] Possible Bug?
Tzafrir Cohen wrote: On Thu, Jul 06, 2006 at 02:27:59AM +0100, Marco Mouta wrote: I'm also surprised there is no faster way to run kudzu than to (yuck) reboot. How about just typing /usr/sbin/kudzu as root from a console? W ___ --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] Sip voip call termination in Nigeria
Pele Zico wrote: Ive looked for cheaper mobile voip pstn calls in Nigeria and the cheapest ive found is about 18p/min or 28c/min. Im looking at providing for cpmanies here cheaper calls comparable to call cards. Can someone give me some ideas as to how i can do this. Can you negotiate prices with a reliable PSTN terminal provider. This is a big project and this inof would help alot Try asking this on the asterisk-biz list. You will have a better chance of getting the answer you are looking for. W ___ --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
