[asterisk-users] Can a BLF show busy only if all devices are busy?
We all know you can monitor multiple devices in one hint, and it shows busy if any device in the group is busy. This is good for a user with multiple devices, but not useful for teams where any person could take a call, like a customer service group. Does anyone know if it's possible to have a hint with multiple devices which only shows busy if every device is busy? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can a BLF show busy only if all devices are busy?
On Fri, Aug 9, 2013 at 2:32 PM, Alec Davis siva...@paradise.net.nz wrote: ** Sounds like you are using queues, you may be able to use the following; In the cases I'm thinking of, we aren't using queues. It's just 2-4 phones and they don't want a queue specifically, so I've just done a multi-device dial to keep things simple. I think I could probably move them to a queue instead. Unless someone else has an idea. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limitation on number of contexts in extensions.conf
On Wed, Jul 24, 2013 at 11:49 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. There probably is a limit, but I don't know what it is. We have many hundreds of contexts and around 80 include files in our main server. My guess is you have an error somewhere. If you show dialplan, does it seem to stop at a certain point as if it loaded only up to a certain file/directory? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE L3 Switches
On Mon, Jul 15, 2013 at 8:40 AM, Eric Wieling ewiel...@nyigc.com wrote: Unless it runs IOS, I don't think most of us would consider that box a Cisco Likely it is a Cisco branded switch with Linksys hardware, i.e. consumer grade stuff. They work well in small business. They have a command line that looks and feels like IOS, though I have no idea if it is or not. Note that there is a confirmed bug with LLDP and auto-VLAN on the SF/SG switches and I haven't heard that they fixed it. If you have phones with CDP, or manually provision VLANs, or don't use VLANs, then no problem. They work great and are reliable. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dongle or extra channel and sip SMS
Something to check out: http://www.kickstarter.com/projects/smush/smart-sms-texting-for-everyone-the-smushbox I'm not affiliated with them at all, but have done business with the company on other things and have always been happy. On Mon, Jul 15, 2013 at 1:57 AM, Chris Bagnall aster...@lists.minotaur.ccwrote: On 15/7/13 3:00 am, bilal ghayyad wrote: I need to be able to send SMS messages for campaign or for specific users, also I need to be able to receive SMS messages and do automatic reply. In my experience, SMS is something best done out of Asterisk. That's not to say that Asterisk can't do it, of course, just that there are providers out there who can give you a nice friendly API for easy integration into your application. This is especially true if you need to send *lots* of messages in a short space of time: simply adding a single mobile device with a single SIM isn't going to cut it - you're going to need a bunch of them, at least. All of those will likely have different numbers, so you're going to have to handle that for receiving messages. Then you have to consider that some networks will charge more to send messages to numbers on the same network vs. a different network, so you might have to separate out your numbers into networks (easy if they've never been ported; more tricky if they have). Based on past projects (in the UK), the cost of multiple SIM contracts, the necessary hardware to connect them, development time, etc., is usually more than the cost of paying a third party with a suitable API x per message to deliver them on your behalf. Kind regards, Chris -- This email is made from 100% recycled electrons -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
No vacation notice, nothing, other than the system auto-replying saying that the ticket will be closed because we didn't have any action on it. Rather distressing for our customers. On Mon, Jun 17, 2013 at 5:22 PM, Gregory Malsack gmals...@coastalacq.comwrote: No. Although Nicolas may have gone on holiday. I just purchased 2 licenses for fop2 a month or so ago. Carlos Alvarez car...@televolve.com wrote: We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need a second opinion on a new phone system deployment
Interesting product that I was very interested in, but the licensing has one huge glaring problem. Be sure to read the FAQ carefully. If your hardware fails and you replace almost anything in the machine, you have to pay for the product again. On Sat, Jun 15, 2013 at 10:42 AM, Michelle Dupuis mdup...@ocg.ca wrote: ... For redundant/failover of Asterisk checkout HAAST at www.generationd.com The HAAST product sits between Linux and Asterisk, monitors for failures etc, and then fails over to another Asterisk box. It effectively creates a low-cost cluster, moving IP's etc to active peer. It runs with most Linux and Asterisk distro's, and avoids the issues of single point of failure. etc. Michelle (generationD) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and forwarding.
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote: Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's fault, or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have myself more confused than when I started. If someone could give me a solid idea of how forwarding works, and a sample of how to send it to a remote extension, and have it *not* come back to the original extension, that'd be great. You said forwarding but described a process that sounds like call transfer. I'm going to assume you mean the latter? We just had a report of this from a customer on their own server. I haven't had time to investigate it. We have confirmed it with Grandstream and Cisco SPA phones, so it's not just Polycom. As far as the atxferdropcall someone suggested, I did try that and then the call is just dropped off into limbo. The caller is left on hold, and the nothing happens on the called extension or transfer-to extension. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] monitoring Asterisk 1.8
Monitor what parts exactly? Right this moment I'm in the process of installing Munin and the Asterisk plugin to monitor channel usage, SIP connections, and the like. The Munin server is running on a separate machine with just the node software on Asterisk. On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] monitoring Asterisk 1.8
It's not quick or simple, but there's decent documentation. I haven't been saving the links I used, so I can't just give you specific places to look, other than the best Asterisk plugin: https://github.com/munin-monitoring/contrib/blob/master/plugins/asterisk/asterisk TIP: Use chmod 755 on the plugin files after you install them. As to installing Munin itself, just start from their web site and get that running. You will then install the Asterisk plugin, create an AMI user for the plugin to connect to, and set the parameters for the plugin to the server IP and AMI account you just created. Right now I'm working on being able to monitor the servers without installing the plugin on the Asterisk box. This will give Asterisk stats only, but no server stats. Again, what specific things do you want to monitor? On Thu, May 9, 2013 at 12:53 PM, motty cruz motty.c...@gmail.com wrote: Thanks for the suggestion Carlos, do you have a HowTo? can you point me to one. I unsuccessfully follow one found using google. I'm using CentOs 6.0 Thanks, Motty On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote: Monitor what parts exactly? Right this moment I'm in the process of installing Munin and the Asterisk plugin to monitor channel usage, SIP connections, and the like. The Munin server is running on a separate machine with just the node software on Asterisk. On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] monitoring Asterisk 1.8
Then you want a queue manager and reporting tool. Usually when people say monitor Asterisk is has to do with the state of the system itself. You should look at http://www.asternic.net and similar products. Munin will tell you channels in use, but not the other stuff you want. On Thu, May 9, 2013 at 1:12 PM, motty cruz motty.c...@gmail.com wrote: Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No early media on 302 redirect via two servers
We have a situation where we get no early media in this call flow: VoIP origination provider Server1 (our server) Customer server Customer phone with call-forward set Server1 to dial the forward-to number Then there is no early media while the forward-to number is ringing. Our server is Asterisk 1.6 and theirs is 1.8. I tried promiscredir=yes and then the calls fail altogether because rather than using the local channel, it makes a SIP call that is not allowed. I don't think that the FORWARD_CONTEXT variable is used in these versions, because setting it doesn't impact the channel selection at all. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
On Tue, May 7, 2013 at 10:05 PM, Satish Barot satish4aster...@gmail.comwrote: promiscredir= yes in sip.conf should help you achieve your requirement. I haven't been able to get that to work in a similar situation, except we are the provider. It results in the new invite being from the CLID of the original caller, and fails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing a sound file during a call
I have a customer who would like to play a series of sound files during a phone call on demand. There would be several played in order during a call. Any simple ideas on doing that without developing a whole web app to do it via AMI? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file during a call
Unfortunately that only plays the file to one side according to the examples, so there's no way for the other person to know when it's done. The caller on the Asterisk server would start the playback, and would need to know when it's done. On Thu, May 2, 2013 at 2:58 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: I think features.conf has what you want under the [applicationmap] setting. They even have an example that would be almost exactly like what you want. From the example: ;testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;;tt-monkeys to the opposite channel Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:Carlos Alvarez car...@televolve.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:05/02/2013 04:53 PM Subject:[asterisk-users] Playing a sound file during a call Sent by:asterisk-users-boun...@lists.digium.com I have a customer who would like to play a series of sound files during a phone call on demand. There would be several played in order during a call. Any simple ideas on doing that without developing a whole web app to do it via AMI? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file during a call
Good point, and that should be usable for the customer. However I'm finding that I can only have about 25 files available to play, and they need 30. Still trying to figure out why that would be, seems like the set command can't parse all 30 possible feature names. On Thu, May 2, 2013 at 3:07 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: Add MOH_Class onto the example and the idle channel will hear music on hold until the playback is complete on the other channel. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:Carlos Alvarez car...@televolve.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:05/02/2013 05:02 PM Subject:Re: [asterisk-users] Playing a sound file during a call Sent by:asterisk-users-boun...@lists.digium.com Unfortunately that only plays the file to one side according to the examples, so there's no way for the other person to know when it's done. The caller on the Asterisk server would start the playback, and would need to know when it's done. On Thu, May 2, 2013 at 2:58 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: I think features.conf has what you want under the [applicationmap] setting. They even have an example that would be almost exactly like what you want. From the example: ;testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;;tt-monkeys to the opposite channel Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:Carlos Alvarez car...@televolve.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:05/02/2013 04:53 PM Subject:[asterisk-users] Playing a sound file during a call Sent by:asterisk-users-boun...@lists.digium.com I have a customer who would like to play a series of sound files during a phone call on demand. There would be several played in order during a call. Any simple ideas on doing that without developing a whole web app to do it via AMI? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file during a call
In case anyone else sees this discussion in the future, the Set(__DYNAMIC_FEATURES) line can't be over a certain length or it stops parsing anything after that. Thanks for the tips, Kevin. On Thu, May 2, 2013 at 3:37 PM, Carlos Alvarez car...@televolve.com wrote: Good point, and that should be usable for the customer. However I'm finding that I can only have about 25 files available to play, and they need 30. Still trying to figure out why that would be, seems like the set command can't parse all 30 possible feature names. On Thu, May 2, 2013 at 3:07 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: Add MOH_Class onto the example and the idle channel will hear music on hold until the playback is complete on the other channel. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:Carlos Alvarez car...@televolve.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:05/02/2013 05:02 PM Subject:Re: [asterisk-users] Playing a sound file during a call Sent by:asterisk-users-boun...@lists.digium.com Unfortunately that only plays the file to one side according to the examples, so there's no way for the other person to know when it's done. The caller on the Asterisk server would start the playback, and would need to know when it's done. On Thu, May 2, 2013 at 2:58 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: I think features.conf has what you want under the [applicationmap] setting. They even have an example that would be almost exactly like what you want. From the example: ;testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;;tt-monkeys to the opposite channel Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From:Carlos Alvarez car...@televolve.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:05/02/2013 04:53 PM Subject:[asterisk-users] Playing a sound file during a call Sent by:asterisk-users-boun...@lists.digium.com I have a customer who would like to play a series of sound files during a phone call on demand. There would be several played in order during a call. Any simple ideas on doing that without developing a whole web app to do it via AMI? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file during a call
On Thu, May 2, 2013 at 5:24 PM, Richard Mudgett rmudg...@digium.com wrote: You can also put dynamic feature group names into the DYNAMIC_FEATURES list. I expected that to have the same limitation, but it doesn't. Works fine, thanks! -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple provider for incoming
Toll free can be moved by a RESPORG in minutes. I know a company who is Asterisk based and does high availability for critical numbers. Let me know if you want an intro. Sent from my iPhone On Apr 30, 2013, at 9:09 PM, Matt Hamilton mistral9...@hotmail.com wrote: Don, Inbound reliability is very important. We don't use toll-free numbers, but we will look into that. I thought porting numbers - not sure about toll-free though - from one provider to the other took days (not technically, but paperwork, etc.) Thanks, Matt -- From: d...@donkelly.biz To: asterisk-users@lists.digium.com Date: Tue, 30 Apr 2013 22:38:44 -0500 Subject: Re: [asterisk-users] multiple provider for incoming If inbound reliability is important, you may be able to accomplish what you want with redundant servers, multiple sip providers and toll-free numbers that can be readily switched between the sip providers. --Don *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] *On Behalf Of *Matt Hamilton *Sent:* Tuesday, April 30, 2013 10:25 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] multiple provider for incoming The process will depend on your provider, of course, but I know some have an option that if they are unable to reach your box, then they can auto-forward to another DID, or to a voicemail box, or to a user-defined function, etc. Forwarding to another DID will/should work for us assuming they are going to be able to do that during a failure on their side. During a recent outage (I think they had some major issues at one of their switches), they were not able to send the calls to our box which was online. Thanks, Matt -- Date: Tue, 30 Apr 2013 20:38:19 -0500 From: wcse...@selbytech.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] multiple provider for incoming On Tue, Apr 30, 2013 at 7:50 PM, David Wessell da...@ringfree.biz wrote: Hi Matt, You can't have multiple providers for inbound traffic. You can have multiple providers for outbound traffic though. Thanks David David, I'm not sure where you got this information, but it's not accurate. I've had multiple inbound and outbound SIP providers for years going to a single box. You just get a separate DID from each provider. Matt, The process will depend on your provider, of course, but I know some have an option that if they are unable to reach your box, then they can auto-forward to another DID, or to a voicemail box, or to a user-defined function, etc. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones
Well the solution turned out to be putting the Asterisk server name in the Proxy field as well as in the server field. Then it properly formatted the SIP registration request. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones
We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's clear that it's registering improperly: [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from 'abc123 sip:abc123@' failed for '68.2.x.x' - No matching peer found Typically of course we'd expect to see: sip:abc123@server We're running the latest available firmware, but it's from 2009. Any ideas on this before we just trash all the older phones? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logging SIP connection status for review
Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within a range, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging SIP connection status for review
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards asterisk@sedwards.comwrote: dumpcap can capture all of the SIP (and RTP) packets into a series of files without a huge performance hit. A cron job can pbzip2 the files and delete if over x days old. That's completely different. We already run a good packet capture system. What I want to see is SIP registration statuses and latency logged about once a minute. We do that now by doing a 'sip show peers like x' and putting it in a text file. I can then correlate issues with times of high latency or unreachable phones. I'd just like to see more reporting and the ability to correlate times and such. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To queue or not to queue...
On Thu, Mar 28, 2013 at 12:55 PM, Gregory Malsack gmals...@coastalacq.comwrote: Here's the scenario~ 150 agents, all are commission based sales reps. 99% of the calls are answered within the first ring. the rest are answered between the second and third ring. Never in my 4 months with the company has a queue call been in the queue more then 20 seconds. Problem~ Several times a week or sometimes a day, the reps will tell me that the same call will be answered by 3 or 4 or 5 reps, and none of them get the inbound audio. Asterisk only shows 1 of the reps actually connecting the call, however the call logs in Eyebeam for all 5 reps, show that they took the call and were connected for a short period of time before disconnecting the call because there is no inbound audio. Which version of Asterisk? Have you looked for solutions to the root problem? I don't run any servers with that many agents, but have never run into issues like this with a few dozen. Large ring groups can become unwieldy and problematic themselves. There's also a limit to how long the entire dial string can be, though I can't remember what that size is. You said everything is on a LAN, but have you looked at the possibility of issues between switches? Can you examine the logs of bad calls and see if the failures happen on a specific switch in the network, or other correlation like that? Do you use VLANs or layer 3 switching? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)
All other phones we work with will auto-answer when we do this: [macro-paging1way] exten = s,1,SIPAddHeader(Call-Info: answer-after=0) exten = s,n,Page(${PAGINGLIST}) exten = s,n, Hangup The SPA phones simply ring. I have verified that Auto Answer Page is set to yes (the default). We've tried a variety of firmware versions and phone ages, going back to an old 942 and new 504s. Any ideas? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)
On Thu, Mar 21, 2013 at 11:58 AM, Optical Phoenix opticalphoe...@gmail.comwrote: Hi Carlos, According to this site, http://community.linksys.com/t5/VoIP-Phones/SPA942-auto-answer-page-intercom-beeping-loudly/td-p/215064the sip string should be Call-Info:\;answer-after=0. I have not tested this yet however. Thanks, that does work. Seems to not interfere with the Grandstream and Polycom phones' operation either. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with 1000 extensions
On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nzwrote: This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Grandstream makes some inexpensive phones that are still very good. Cheapest hasn't been defined yet. What's the budget? Is there existing networking at these locations? Will you need switches? PoE? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error to install Asterisk
I'm going to make an observation here that may upset you, and I don't mean it to, but it's fact. If you are so unfamiliar with Linux, you will have a bad time managing Asterisk servers. You really need to know how to use the OS before you can learn to manage services running on it. I strongly suggest one of the all-in-one Asterisk variants like AsteriskNOW. There is simply no way to run a production server without having to do systems management regularly. On Wed, Mar 6, 2013 at 3:01 AM, termo termosel fermit...@hotmail.comwrote: Hi, this is the outpu to df -h command: root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficherosTam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0668M 668M 0 100% /rofs tmpfs1006M 44K 1006M 1% /tmp none 5,0M 0 5,0M 0% /run/lock none 1006M 100K 1006M 1% /run/shm Jordi -- From: fermit...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 5 Mar 2013 17:40:32 + Subject: Re: [asterisk-users] Error to install Asterisk Hi, Ok, tomorrow I will send the output when I will be in the office! Thanks! From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Tue, 5 Mar 2013 16:11:01 + Subject: Re: [asterisk-users] Error to install Asterisk On Tuesday 05 March 2013, termo termosel wrote: Hi, when I try to install Asterisk 11.2.1 the console return error which it tells: /usr/bin/ld: final link failed: No space left on device and the process exits installation. How can I solve this problem? Tmp folder is empty. Thanks,Jordi Try entering this command: # df -h and paste the complete output in a message. This will show the amount of space used and remaining on all filesystems, in human-readable notation (i.e. it will automatically select the units: bytes, kilo, mega, giga or terabytes, so as to get a sensible figure). You'll almost certainly have to move some files out of the way. Have you got, or can you get, a USB external HDD; which either already has a Linux ext4 file system on it, or contains only sacrificial data? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error to install Asterisk
On Wed, Mar 6, 2013 at 10:02 AM, Gertjan Baarda gertjan.baa...@gmail.comwrote: Couldn't agree more, Carlos. But then again, haven't we all started this way? ;-) The best way to understand Linux is learning the hard way. After all, it takes a genius to understand the simplicity of Linux. If you're going to learn Linux, then learn it, not via some service running on it. It's clear in context that the original poster believes that he can install and run Asterisk without knowing the OS. This is obviously not true. If it's going to be someone's production server, that is scary. It also has led to many ASTERISK SUCKS! discussions I've had because there were problems at the OS level that made the Asterisk server unreliable. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What would cause a drop between two asterisk systems?
On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.comwrote: We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. IAX is buggy. We've never seen a reliable system using it. We've given up on it. I'd try SIP. Easy to do, no real reason not to. Check all of the networking involved. Leave a ping test running between the systems constantly, then see if it dropped packets when you get a dropped call. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / check / tool
On Mon, Feb 18, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote: A normal dialplan reload command would echo no warning or something similair. The duplicated extension will cause an error. Something like cannot add extension in line X because it already exists. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk calls between 2 private networks
On Thu, Feb 7, 2013 at 10:26 AM, Frank fr...@efirehouse.com wrote: AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. Adding more points of failure and more devices to maintain without any real benefit is always the wrong thing to do. IAX is also flaky as hell. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd question - Give me your opinion please
On Tue, Feb 5, 2013 at 11:43 AM, Ira i...@extrasensory.com wrote: Personally I think I'd purchase a number of used servers so if one dies, you have a backup or three. 25 lines does not need a modern processor and it seems silly to spend money for a warranty when you could spend less and have a number of spares. I completely agree with this. I'd buy several refurb/used machines rather than one new. We buy all of our HP refurb servers from these guys: http://www.nautilusnet.com/ alb...@nautilusnet.com -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect a POTS robo alert dialer to asterisk with email notification
On 3/2/13 4:59 pm, David Smiley wrote: I finally found the perfect solution for me:http://www.amazon.com/La-** Crosse-D111-101-E1-WGB-**Wireless-Monitor/dp/** B0081UR76G/ref=dp_ob_title_defhttp://www.amazon.com/La-Crosse-D111-101-E1-WGB-Wireless-Monitor/dp/B0081UR76G/ref=dp_ob_title_def The device is $69, plus $10/month for alerts. And I get to monitor the temperature online, which is a great bonus. I use these devices to monitor everything from server rooms to my home freezer and cigar humidor (they also have monitors that do humidity). This is the best solution to the problem, not a voice system. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd question - Give me your opinion please
If you have the budget for two machines, run all services on one and keep the other for a hot backup. Rsync the configs nightly. I'm guessing that spare parts/repairs are far away from where you will be? On Mon, Feb 4, 2013 at 9:11 PM, Jared Baxley jared.bax...@gmail.com wrote: Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance. Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards. Analog extensions WILL Hit a Surge Gate before the cards, and as much precaution on grounding protection and power protection is being taken as possible. The cards WILL BE PCI not PCI-e (They are being donated) A New Dell Power-edge Server will be acquired for the PBX HERE IS MY QUESTION Would you purchase a NEW TOWER Server with PCI slots to accommodate the cards, OR Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper Server just for the analog extensions, I'm torn... The ease of management of one server, or the isolation of analog extensions scattered through the jungle on it's own server. Opinions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send record file to email
On Fri, Feb 1, 2013 at 8:35 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 01 February 2013, Bob Kyeyune wrote: Hello; how do i embed and send the recorded file to email automagically exten = _1XXX,3,MixMonitor(${CALLFILENAME}|b|/usr/sbin/wav2mp3 ${CALLFILENAME} ${peeremail} ${EXTEN} ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} ) Here's an example of how we record and e-mail. You will need to download the sendEmail binary. [conference-record] exten = s,1,Answer exten = s,n,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/TelEvolve-conf-${UNIQUEID}) exten = s,n,Set(MEETME_RECORDINGFORMAT=wav49) exten = s,n,Wait(1) exten = s,n,MeetMe(televolve,cDr) exten = h,1,System(/usr/sbin/sendEmail -t car...@televolve.com -f notificati...@televolve.com -u Conference call recording -m Conference call ${UNIQUEID}) -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 9:26 AM, Adam Moffett adamli...@plexicomm.netwrote: Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen at? I don't think I've seen anything that has a register command, but lots of devices can get a check your config or reboot command via SIP NOTIFY. If you can notify, you can call. This fixes nothing other than refreshing NAT if that's involved. I'm more wondering why the peer is unregistered but we still expect to communicate with it. Other than a network problem or the device being unplugged...neither of which could be fixed from the server. I have a feeling that some people in this discussion have a lack of understanding about the SIP protocol and the underlying networking that could affect it. The original post failed to say whether this was on a LAN without routing, on a LAN with routing, or a WAN. Each of those could result in totally different results and solutions. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (SOLVED) Call parking in a multi-tenant system
I figured I'd follow up on this in case anyone else cares. The documentation is simply awful and it took a lot of experimenting to make it work. In features.conf for each company: [parkinglot_televolve] parkext = 700 parkpos = 701-720 context = televolve#parking parkinghints = yes parkingtime = 75 courtesytone = beep parkedplay = both findslot = next Include the context declared above in extensions.conf under the dial context. include = televolve#parking In the defaults for the customer's sip.conf file add: parkinglot=parkinglot_televolve NOTE: The parkinglot_ in the name is required! On Tue, Jan 15, 2013 at 2:08 PM, Bakko asannu...@gmail.com wrote: Hello, from 1.6.2 version, Asterisk suport multi-tenant parking Look at features.conf for a example. Regards El 15/01/2013 15:58, Carlos Alvarez escribió: We use Asterisk as a hosted PBX. We've had a couple of requests for parking, but none of the documentation shows any way to make it aware of contexts or otherwise make it multi-tenant. Have I missed something and does anyone know how to make this work? Would be on Asterisk 1.6 for now, 1.8 some time soon. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There is. If you build a reliable network, the phones will simply never have a problem. We've got customers with phones that have never lost contact for years. Re-registering is just a crutch for a network defect. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quoting error with gotoiftime
I'm getting the following error, and none of us can figure out why: [Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting $end; Input: = ^ Here is the code that generates it: [scottsdale#queues-account] exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue) exten = s,n,Goto(scottsdale#queues-closed,s,1) exten = s,n(queue),ExecIf($[${prefix} = ]?Queue(azenglish):Queue(azspanish)) exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage) exten = s,n,Voicemail(@scottsdale,s) exten = s,n,Hangup Here is the rest of the call progress surrounding it, which seems to be working anyway: -- Executing [2@scottsdale#queues-aax:1] Goto(SIP/televolve-1-1c7d, scottsdale#queues-account,s,1) in new stack -- Goto (scottsdale#queues-account,s,1) -- Executing [s@scottsdale#queues-account:1] GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in new stack -- Goto (scottsdale#queues-account,s,3) [Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting $end; Input: = ^ [Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables -- Executing [s@scottsdale#queues-account:3] ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish)) in new stack -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:20 AM, Eric Wieling ewiel...@nyigc.com wrote: Looks to me like ${prefix} contains nothing but two quotes. Which is as it should be unless they choose the Spanish option, but yeah, maybe that's what is choking Asterisk. We do this: exten = _X.,n,Set(prefix=) ;Initialize variable used in scottsdale#queues-aax Then in the AAX if they choose 5: exten = 5,1,Set(prefix=s-); SPANISH That way we just carry along their preference for Spanish throughout the system. Perhaps I need to initialize the variable as e-, but then would have to rename everything. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote: What version does the error occur on? I suspect more recent versions of Asterisk removes extraneous quotes. This is in 1.8. Danny's test does support your theory. It looks like the var is being set as the quotes, rather than empty. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:31 AM, Danny Nicholas da...@debsinc.com wrote: Where possible you should have a VM to try these things as needed. Where not, it isn’t too difficult to duplicate the contexts and do something like this [default] I do have a test VM, but I also have a maintenance window for this customer later tonight for other things, so I was being lazy. Poor excuse. Tested it, and it works, thanks guys! -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to assign the button on the IP Phone to a feature?
On Thu, Jan 24, 2013 at 8:03 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; Using Cisco IP Phones: How I can assign a button for a function. For example, if we pressed on this button, then we need to pickup the call from the group. Which model line? The SPA series, or the 7900 and similar? They are completely different. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT Solution
On Thu, Jan 24, 2013 at 12:52 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Polycom also has DECT stuff. I doubt it will come cheap. http://spectralink.polycom.**com/dect_communications/index.**htmlhttp://spectralink.polycom.com/dect_communications/index.html Not cheap, but this is the solution for large installations (over say 10 or 20 handsets or big spaces). Reliable, easy to work with. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a need to secure RTP ports?
On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk wrote: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). My question is - what are the vulnerabilities in this scenario at my end? I suppose some man-in-the-middle or eavesdropping attack is always a possibility - but that aside, is there anything that will attack RTP ports on Asterisk when there are no SIP ports open? I was looking into installing fail2ban - until I realised that there is no SIP port exposed for an attacker to poke at. I've been working in IP telephony for about ten years. I've never once heard of any attack on the RTP ports. While you can never say anything is impossible there's simply nothing listening on those ports. It's probably possible to have a DOS attack where someone starts sending RTP to all of your ports and they would interfere with a call, but they couldn't do more than that. That could work if your router has full cone NAT and a lot of other things fall into place. Still kind of out there as a real threat. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, Digium phones, and voicemail.
On Tue, Jan 22, 2013 at 1:48 PM, Frank fr...@efirehouse.com wrote: That worked, thank you. Is there a way to program the keys of the Digiums D70 from asterisk ? Or does everything needs to be done on the phone itself ? The whole point of the Digium phones is their tight integration with Asterisk. You need the Digium phone module, and you really need to read the documentation as this is well covered. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk voicemail minimum length / silence settings
On Tue, Jan 22, 2013 at 4:22 PM, asterisk users ast4...@gmail.com wrote: What are the right settings for this situation? We've used the following settings system-wide for about nine years without one complaint or known issue: [general] format = wav49|pcm maxsecs = 360 minsecs = 4 skipms = 3000 maxsilence = 3 maxlogins = 3 maxgreet = 120 maxmsg = 50 silencethreshold = 128 operator = yes sendvoicemail = yes usedirectory = yes forcename = yes forcegreeting = yes saycid = no review = yes nextaftercmd = yes tempgreetwarn = yes Some of these things may be deprecated depending on the version you're using. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk voicemail minimum length / silence settings
On Tue, Jan 22, 2013 at 4:43 PM, Don Kelly d...@donkelly.biz wrote: But with max silence at 2 seconds, couldn’t someone leave a 30-second message, pause for a couple seconds to gather their thoughts or dig up a phone number, and get hung up on? Two seems short, but nobody has complained about our setting of three. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Capture queue agent drop and put caller back in queue
On Mon, Jan 21, 2013 at 11:03 AM, Mitch Claborn mitch...@claborn.netwrote: How can I accomplish my goal? http://lmgtfy.com/?q=battery+backup -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recrding calls
On Fri, Jan 18, 2013 at 6:25 PM, Joseph syscon...@gmail.com wrote: I would like to outgoing/icoming calls and email the files. This is what I have: ... exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP}) exten = _7.,n,Monitor(wav,${**CALLFILENAME},m) ... How do I email these file? This is how we do it: exten = _1NXXNXX,1,Set(recordfilename=/var/spool/asterisk/monitor/${EXTEN}-${TIMESTAMP:0:8}${TIMESTAMP:8}.WAV) \exten = _1NXXNXX,n,MixMonitor(${recordfilename},b) exten = _1NXXNXX,n,(dial here or whatever) exten = h,1,System(/usr/sbin/sendEmail -t u...@domain.com -f p...@domain.com-u Call recording for ${recordingfilename} -m There is a new call recording. -a ${recordfilename}) Google the sendEmail app and download it, very useful for a lot of things. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recrding calls
Yeah, sorry, that example was from one of our older servers. On Fri, Jan 18, 2013 at 8:08 PM, Warren Selby wcse...@selbytech.com wrote: Around version 1.4 or 1.6, TIMESTAMP was phased out and replaced with STRFTIME. See this page for details on how to properly generate a timestamp: http://www.voip-info.org/wiki/view/Asterisk+func+strftime On Fri, Jan 18, 2013 at 8:46 PM, Joseph syscon...@gmail.com wrote: On 01/18/13 19:27, Carlos Alvarez wrote: On Fri, Jan 18, 2013 at 6:25 PM, Joseph [1]syscon...@gmail.com wrote: I would like to outgoing/icoming calls and email the files. This is what I have: ... exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP}) exten = _7.,n,Monitor(wav,${**CALLFILENAME},m) ... How do I email these file? This is how we do it: exten = _1NXXNXX,1,Set(**recordfilename=/var/spool/** asterisk/monitor/${EXTEN}- ${TIMESTAMP:0:8}${TIMESTAMP:8}**.WAV) \exten = _1NXXNXX,n,MixMonitor(${**recordfilename},b) exten = _1NXXNXX,n,(dial here or whatever) Thanks Carlos I'm just concentrating right now on ${TIMESTAMP} variable but is is not working: I have: exten = 11,n,Set(recordfilename=/var/**spool/asterisk/monitor/${** EXTEN}-${TIMESTAMP:0:8}${**TIMESTAMP:8}.WAV) exten = 11,n,MixMonitor(${**recordfilename},b) and the file name I got was: -11.wav Why I'm not getting any timestamp? -- Joseph -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call parking in a multi-tenant system
We use Asterisk as a hosted PBX. We've had a couple of requests for parking, but none of the documentation shows any way to make it aware of contexts or otherwise make it multi-tenant. Have I missed something and does anyone know how to make this work? Would be on Asterisk 1.6 for now, 1.8 some time soon. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] block one number in incoming calls
So I'm not the only one who uses the monkeys as our place to send bad calls to. -- Sent from my iPhone On Jan 14, 2013, at 10:02 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Monday 14 January 2013, Salaheddine Elharit wrote: i think i didn’t explain correctly may question i revive a lot of calls from this number _0666XX and i wants to block it to call my number 520xx . Use something like Exten = _520X./0666XX,1,Answer() Exten = _520X./0666XX,n,PlayBack(tt-monkeys) Exten = _520X./0666XX,n,HangUp() Now when a call comes in from 0666XX to _520X. they will get monkey noises. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] block one number in incoming calls
On Mon, Jan 14, 2013 at 10:48 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Monday 14 January 2013, Carlos Alvarez wrote: So I'm not the only one who uses the monkeys as our place to send bad calls to. I actually thought of using a sample from a well-known song by Australian singer Kevin Bloody Wilson, but decided that might be a bit *too* offensive. Besides which, the monkeys sample was already there . For the really obnoxious problem callers, I typically find out their corporate phone number and just forward all the marketing calls back to them system-wide. I did that with the car warranty people. I wish I could have legally recorded them, that could have been fun stuff. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How often to restart Asterisk...
On Fri, Jan 11, 2013 at 2:06 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until there is a problem? This box had 119 days of up time on the Asterisk process. I have a client that I installed an Elastix instance on and the last time I checked it, it was up to almost 500 days of up time without an asterisk restart. I've had boxes run for years, and others have problems in a month or two. I have a general practice of having a reboot cron job on critical servers at 3am on Sunday. Our customer SLA allows for a maintenance period during this time. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On Thu, Jan 10, 2013 at 10:18 AM, Matthew J. Roth mr...@imminc.com wrote: You already have all of our addresses. Please unsubscribe us. It's the only way to partially redeem yourself on this list. Do you really want to be pissing off some of the most active Asterisk users? You've already guaranteed that a lot of us will actively tell people not to do business with you. I've had the opportunity to warn two potential users in the last year. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote: It really didn't bother me as much as reading all the posts but that's just me...now back to Asterisk issues :) Sorry to add another, but for me, the main point is that this activity speaks to the character, ethics, and trustworthiness of the company doing it. We all have spam filters. I just also add the company to my do not buy/do not recommend list. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
Hopefully it's not, What is the best DID provider for Asterisk... On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro stot...@totarotechnologies.com wrote: So what asterisk issue do you have? Let's fix it. On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler rwhee...@artifact-software.com wrote: That does not solve any asterisk issue that I have. On 10/01/2013 1:32 PM, Carlos Alvarez wrote: On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote: It really didn't bother me as much as reading all the posts but that's just me...now back to Asterisk issues :) Sorry to add another, but for me, the main point is that this activity speaks to the character, ethics, and trustworthiness of the company doing it. We all have spam filters. I just also add the company to my do not buy/do not recommend list. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On Thu, Jan 10, 2013 at 3:09 PM, C. Savinovich c.savinov...@itntelecom.comwrote: Although people complaining of spam may be valid, the one part of complaining about spam that bothers me, is that some people should look at themselves in the mirror and ask the question aloud Why does it bothers me to see another Asterisk professional compete with me for jobs?, Why do I get angry at seeing someone else's solicitation for services I too provide? In reality, anyone who solicits customers in this list does a folly thing because this list has more asterisk consultants than customers. This list is the equivalent of a Home Depot parking lot full of construction workers looking for a job. A lot of people on this list are potential customers of the company in question. I doubt that most of us compete with them. We used to be their customer. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing music through VoIP handsets while on hook
This is something I've seen with some key systems and PBXs. When the phones are on-hook, they can play music throughout the office instead of having an overhead speaker system do it. Never heard of it being done with VoIP, but figured I'd ask if anyone else has. I don't see any way to do this on any phones I've looked at. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing music through VoIP handsets while on hook
On Thu, Jan 10, 2013 at 6:42 PM, Christopher Harrington ch...@acsdi.comwrote: Wow, that seems wildly bandwidth inefficient. Is it possible to do multicast VoIP? Depends on whether the phones are local to the server. Unless you're looking at hundreds of phones, a 100MB network running 80k to every phone wouldn't even notice. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
I don't buy from spammers. Buying email lists may be working for you, but some people will never do business with you because of it. We cancelled our account with you because even as a customer we were subjecting to your unscrupulous marketing. -- Sent from my iPad On Jan 9, 2013, at 7:10 PM, Jai Rangi jpra...@didforsale.com wrote: Don, I have removed yours right away. Yes, I agree, But just like any company we have purchased/collected email from different source. Also just like any company we are not perfect, we make mistakes. -Jai Rangi On Wed, Jan 9, 2013 at 5:57 PM, Don Kelly d...@donkelly.biz wrote: Jai, ** ** It should not be necessary for me to remove my email address from your list. It should not be on there to start with—we do not have, and have never had, a relationship that justified you sending me email. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jai Rangi *Sent:* Wednesday, January 09, 2013 7:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] DIDForSale spam ** ** Guys, Since I am attached to did for sale: My apology to every one who received the DIDForSale 2012 Achievement email and you hated it. As a asterisk user my question will be. If some xyz company send you a so called spam email, what made you think that you should spam the mailing lists. I am sure we all get lots if spam emails every day. If you really got some time and talent, why don't you write some good tips and tricks about asterisk. Long story short We have a link where you can unsubscribe your email for any further communication. http://www.didforsale.com/unsubscribe.php or Send me your email address I will personally take care of that and will remove your email. This will take less than 5 seconds. I am sure there will be lot of arguments on why you should that and all. I will refrain myself on any further unproductive communication. Happy new year to you all. On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:*** * +1 here. On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote: On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote: What were the senders IP(s)? Will have to look it up when I get home. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have gotten hit with this twice so far. in March and Today: Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com 3/8/12 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com 1/9/13 UGH, when I asked in March where he got my email he said: Hi Chris, We got your contact from the Internet. Let me know the good time to talk about this in detail. Thank you, -Rohit Dhaka Obviously by harvesting these lists. I received 2 myself. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] Paging unit suggestions
On Mon, Jan 7, 2013 at 1:10 PM, Doug Lytle supp...@drdos.info wrote: We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to replace both the system and hopefully move to a IP paging system, but wanted to reuse the current speakers. If the amps are good, you could just drive them from a cheap phone with a regular headset jack. Set it to auto-answer, send the paging extension to it. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon SIP trunking Field Trial
On Fri, Jan 4, 2013 at 11:18 AM, Eric Wieling ewiel...@nyigc.com wrote: Trust me, Verizon doesn't really provide support.What they will do is tell you something different (often conflicting stuff) when you send in a ticket.One time they tell us the From must be in e.164 format, other times they say it does not.We asked for an updated Interop guide weeks ago and they have not provided us anything. We have been with VZ SIP for years so I wanted an updated interop guide so we can point them to it when they tell us something which conflicts with their docs. Don't get me started on trying to upgrade our service with them.. Sounds like the same huge effort it takes to work with Qwest/Centurylink, and in the long run we found it simply isn't worth it. The few benefits of working with an RBOC are countered by the many drawbacks of working with an RBOC. Also we recently acquired a half million minutes/mo from a company who was tired of dealing with Qwest SIP. They said the same thing I said above. I suppose the point of what I'm saying is you should really think about what you think you will gain from a relationship with them, and whether all this is worth it (all this means now and how their attitude will affect you forever). -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon SIP trunking Field Trial
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com wrote: Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? It may be too late for this, but in working with another RBOC who didn't want to deal with Asterisk, I just asked what they do support, and modified the headers sent by Asterisk to claim that it was one of the devices on that list. Done. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com wrote: And I started communicating with a 2400 baud modem so trimming was a necessity and a requirement of friendship. Bah, spoiled kids. Mine was a 110 baud acoustic. I think the Will Asterisk run on a Rasberry Pi thread the perfect example of why this list is dying. The number of questions posted here that are easily answered with a search or which are far too basic and open (how do I make Asterisk work) is very high these days, and that does kill a list. A lot of us are interested in helping people who help themselves, and solving complex problems. I've seen many tech lists die off when people stop trying to help themselves and ask intelligent questions. As to top-posting, another example of when I think it's generally acceptable is people using tablets. I have found no way on either my iOS or Android tablets to quickly/easily post in the traditional manner. If I'm faced with spending a few minutes carefully trimming a useful reply or just not posting it at all, I'm likely to choose the latter if I'm on a list that says absolutely never top post. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto ban IP addresses
On Wed, Jan 2, 2013 at 3:49 PM, Frank fr...@efirehouse.com wrote: Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;** tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? http://www.fail2ban.org/wiki/index.php/Asterisk -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
There's nothing wrong with the list. The current whining will die off and things will be back to normal shortly. Meanwhile I was tempted to bin the entire thread then realized there's some funny human psychology to be laughed at. Top posting to make a few people crazy. Normally I wouldn't. On Sun, Dec 30, 2012 at 7:37 PM, James Mortensen james.morten...@voicecurve.com wrote: I have an idea! Instead of arguing over whether or not top posting or bottom posting is the way to go, something that obviously no one will * ever* agree on, why not move to Google Groups instead (or something similar to Google Groups). When I post to Doubango's list, it's easy, there's no top or bottom posting wars, it just works. In fact, in a thread, Google Groups usually drops you right to the most recent message, so the people who like top posting can still see the most recent message while the bottom posters will still see the bottom posting format. It's either this, or we can sit and watch intelligent people continue to degrade one another and argue over something with no agreement in site. :) When I mentioned this before, someone from Digium said this will never happen, and it's unfortunate. Maybe they just like to see people bicker and argue. If there's a better alternative to Google Groups, or a way to set preferences in the mailing list so that everyone is happy, maybe that's something that could be done? James On Sun, Dec 30, 2012 at 6:30 PM, Ron Wheeler rwhee...@artifact-software.com wrote: On 30/12/2012 11:13 AM, Patrick Lists wrote: On 12/30/2012 04:26 PM, Ron Wheeler wrote: I participate in a lot of lists and top posting is now the norm since people want to see quickly if the message is worth reading. Isn't it a bit of a stretch to extrapolate your experience with your lists to top posting being the norm? I am subscribed to several lists and bottom posting, proper trimming and commenting inline is the norm there. Actually the norm is determined by the list rules. If the list rules say one must use bottom posting then one should use bottom posting. If someone does not like that then don't subscribe, find another source to ask a question (the forum, LUG, hire a consultant) or just bottom post. Questions come before answers. Answers come after questions. -1 against changing rule #5. Regards, Patrick Not really enough time in the day to keep track of different rules for all the forums. I am more concerned about content than form. As long as the questions get answered, I can figure out where it is but it is a PITA to scroll down through an e-mail to find out that there is nothing there worth reading. I get over 100 e-mails per day that make it through my filters. I like to read the content as soon as it pops up rather than searching for the text. Ron -- __**__** _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Cisco 887M
On Thu, Dec 27, 2012 at 9:10 AM, Edwin Quijada listas_quij...@hotmail.comwrote: Hi! I am installing asterisk with my ISP but he give me a Cisco 887M router to use for SIP conection. My problem is that I dont know how to link Asterisk with this device because I dont have user/pass to use. Anybody has a cluee to use CISCO 887M with Asterisk ? What are the symptoms of the problem? One way audio? No audio? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.cawrote: This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Boxing day??? Seriously? There's a holiday for people who beat each other up? TIL. But anyway the best way to test time-based rules is on a VM that has a copy of your configs, and just change the time. You can easily run a small VM to accomodate a copy of your main server on almost any computer. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding / Follow-Me on PRI
On Thu, Dec 27, 2012 at 2:41 PM, Barry D. Hassler barry.hass...@gmail.comwrote: Friends, Curious if others have run into this scenario, and can shed further light on it. I am working with an installed base of systems using PRI circuits from several carriers, and the symptoms I relate occur across the board. We have encountered it, and simply told the carriers to stop blocking it or lose the business. All but one did it, and we dropped their services. Don't know that there's a good work-around otherwise. Is there a reason you don't just go all SIP, where 98% of the service providers will accept any CLID? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change phone display from queue calls
We are trying to set up a system where the calls from the queue show a specific name or number on the phone. The calls would come into one of a few dozen DID numbers, each one for a specific company. The agent needs to know which company the call is for and answer appropriately. I've done a lot of this in dialplans but haven't found a way to do it in a queue. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need qualifications of SIP trunk providers
On Fri, Nov 30, 2012 at 7:10 AM, Daniel - Asterisk earohua...@gmail.comwrote: Thank you Carlos, What does mean 'por-out'? I'm expecting 1 min/month in out. Port out means a number was ported to another carrier. 10k minutes is not huge but a decent number that should get you a reasonable rate with the carrier of your choice. Vitelity is a good fit for that size. I can't say they are better than the others because I haven't used them, but we had a few hundred DIDs and probably did 50k minutes with them at one point. As we've grown we have moved to others (we're around half a million min/mo now). -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need qualifications of SIP trunk providers
On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Vitelity is reliable and decent, but no phone support. Have not used the others. Oh also if you lose a number on Vitelity to a port-out, they won't know and won't stop billing you for it. What's your expected volume in/out? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] watchdog like functions
Switching to SIP is likely your best solution. IAX is buggy. Always has been, and I'll bet always will be. On Tue, Nov 20, 2012 at 7:34 PM, asterisk asterisk aster...@ck-lee.comwrote: I wish to ask if there is way to keep IAX trunk connection up. I have a small server on Xen VPS but notice that my IAX trunk drops after some time. I understand there is cron job to function as sip watchdog. My asterisk is 11.0.1 Thanks for suggestions. CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3G Quality
On Wed, Nov 14, 2012 at 3:15 PM, Roy Abshire r...@coopvr.com wrote: Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. VoIP does not work reliably over 3G. I've had best success with g.729 but it's still not great. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3G Quality
On Wed, Nov 14, 2012 at 3:30 PM, Roy Abshire r...@coopvr.com wrote: Ok, what about 4G, I tried it with my 4G ATT Hotspot too. I get 4 full signal and the call goes through but keeps cutting out or dropping. I'm using the ATT Elevate with a Nokia N97. SIP connects fine, I can make and receive calls and when I check the console it connects using ilbc but we have a hard time hearing each other clear and it drops unexpectedly without hanging up the other side. I wouldn't expect any cellular connection to pass VoIP reliably. Think about it. The phone call is on the same airwaves but is prioritized. The data gets the lowest priority. Again, all my testing has left me using only g.729 when I really need to do VoIP over cellular, but it's never been perfect. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3G Quality
On Wed, Nov 14, 2012 at 3:34 PM, Roy Abshire r...@coopvr.com wrote: G729 requires a paid license right? If I wanted to test it using G729...do I still have to buy a license for it? My goal is to use my cell phone for Text Only and pay $15/month and use my ATT WiFi Hotspot for VOIP calls wherever I go because I rely on my VOIP for business. It's a $20 license, dirt cheap. If your goal is to simply save the $45/mo it costs for a full unlimited plan, you're wasting a lot of time and energy. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3G Quality
On Wed, Nov 14, 2012 at 4:20 PM, Roy Abshire r...@coopvr.com wrote: My goal is to always be connected to my VOIP system using 3G or 4G. I'm still going to have to pay the ATT Data only $50/month plan for 4G and I get 5gb a month which is plenty. Why would you always want to be connected to the VoIP system? Just do concurrent ringing to the cell number like almost everyone does. Yeah, I wish the VoIP over cell data fantasy worked also, but it really doesn't, as far as I've seen and heard. Particularly while moving. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3G Quality
On Wed, Nov 14, 2012 at 5:22 PM, Roy Abshire r...@coopvr.com wrote: Believe me, there is a method to my madness that I didn't want to get into but here it goes. Quite often, telling people the core problem can lead to better solutions. I assumed you wanted mobile, but fixed cellular could work for this. I want to get this to work reliably even with low quality and bandwidth so I can install VOIP Phones at my vacation rental properties that can only get 4G or low speed Satellite with only 128k upload. 128 upload is no problem, but latency on geosynchronous satellite is horrible. Pretty much not usable for voice. LEO satellites like Iridium and Inmarsat work, but are VERY expensive. Have you checked out the new Dish network satellite internet service? I haven't used it, but the specs look great and the price is cheap. I went up and stayed at one of the properties and bought my ATT 4G Elevate device just to test it out over VOIP at the cabin and I got 2 bars and my Nokia connected and registered over SIP...but calling was garbled. What you need is an external antenna, possibly a directional one if you know the general direction of the cell tower. Wilson makes some great amplifiers and antenna systems. I use a Wilson amp on my boat with a tall omni marine antenna, and it lets me make calls over VoIP where there would otherwise be no signal (they are choppy but usable). There is some controversy on the best CODEC, but my experience is that g.729 is best on low-quality connections. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static on calls - v1.8.15.0
On Thu, Nov 8, 2012 at 9:34 AM, James Lamanna jlama...@gmail.com wrote: Hi, I'm testing out a server with asterisk 1.8.15.0 on it. I'm experiencing static occurring on almost 90% of calls on this particular server. All test phones are using SIP, and calls to/from PSTN servers are delivered using IAX2. I have other production servers running 1.4.x that do not have this issue that use the same PSTN connections. I haven't seen any ethernet errors or anything like that. Load is minimal since this is still a test server. The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s. I'm sort of baffled as to where to start looking for the root cause of this issue, but it appears to be isolated to only this machine. I would recommend you try SIP and lock out all CODECs on both sides except ulaw. See what happens. We've had this problem both because of CODED mismatch and just due to IAX's many many bugs. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] forwarding all calls to cells
On Tue, Nov 6, 2012 at 5:33 PM, Noam Birnbaum n...@maccentricsolutions.comwrote: Hello everybody, A client wants to install a FreePBX infrastructure, but have all calls forward to their cell phones rather than buying VoIP phones. They would be doing SIP trunks over a Comcast business line. Probably maximum 6 simultaneous calls. Any gotchas we should warn them about? What a waste of effort, bandwidth, and money. There are a dozen services out there that can do this far more efficiently and for a lot less money. You will be using bandwidth in both directions and double the channels (one call in, forwarded call out). It will increase the latency by a lot and the chance of call quality problems is huge. Call in over a cheap cable connection, call out on the same, then over a cell. Concurrent ringing with cells is popular and makes a lot of sense since you primarily typically answer on the desk phone. If it's just forwarding, then just use a forwarding service which will cost less and not introduce the extra cable network layer. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multitenanat third party app
Indeed this is getting ridiculous. This person also called me (!!) for some free consulting after I had posted the answer a few days ago. NOTE: We aren't going to engineer your system for you! We as a group will provide help and some basic code to get you started. If you don't know how to start working with the fully working stuff I provided already, you're not ready to deploy a system this complex. On Wed, Oct 31, 2012 at 2:59 PM, Mitul Limbani mi...@enterux.in wrote: Stop asking same questions !!! On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote: Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. There are multiple ways to do this. One way is the Local dial. We have done this for companies who are different entities but want to do 3-digit dial. Dial(Local/101@company_a#extensions,25) Where we assume you have a context like: [company_a#extensions] exten = 101,1,Dial(SIP/company_a.${EXTEN},25) Another way is to simply do an include for the other company's extension context. However that requires that you not duplicate the extension numbers between the contexts/companies. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
I am attempting to send this again. The mail processor is interpreting the Asterisk commands in my message as mail processor command and bouncing the message. That's why where is junk before many of the lines below. On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. There are multiple ways to do this. One way is the Local dial. We have done this for companies who are different entities but want to do 3-digit dial. ... Dial(Local/101@company_a#extensions,25) Where we assume you have a context like: ... [company_a#extensions] ... exten = 101,1,Dial(SIP/company_a.${EXTEN},25) Another way is to simply do an include for the other company's extension context. However that requires that you not duplicate the extension numbers between the contexts/companies. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I’m looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I’d like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I’ve been able to find with a 48 port capacity are these two: I have a deployment of 96 analog ports using a Digium T1 card ($500 on eBay) and Rhino analog channel banks (also cheap on eBay). We have extremely high reliability from this configuration. In fact, other than the normal analog annoyances like occasional echo, they are rock solid. Are you doing this instead of VoIP phones for cost reasons? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen jkil...@allamericanasphalt.com wrote: ** Cost and ease of deployment, yes. At this specifc location we are currently using Centrex lines (ATT hosted) and are looking for a way to move into something cheaper without throwing away the existing phones. I like the idea of using a channel bank – I’ll look into that as an option as well. You should be able to also connect the Centrex lines to the channel banks, I believe. I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 2:14 PM, Christopher Harrington ch...@acsdi.comwrote: On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote: I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. Out of curiosity, would you mind sharing that with us? The phone? Grandstream. They have people who love them and hate them, but so far we're pretty happy with them. A GXP2124 is on my desk right now, and I have access to any phone I want. I like it. There are low-end models still with a display for under $60. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 2:34 PM, Justin Killen jkil...@allamericanasphalt.com wrote: ** ** ** I think if we were to go to VoIP phones, one thing that we would have to consider very highly in a phone would be that they have VLAN settings and a built-in Ethernet hub/switch so that we can just inject it into the user’s computer LAN connection. The cost and time of rewiring some of these locations is not something we’re comfortable with doing. The Grandstreams do that. Most phones do. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail to text for Asterisk
A customer has asked us to provide that feature. I know there are a few methods and products out there, but I haven't paid attention in a while. It is for about 300 users, and we'll consider open as well as paid-for products. We would prefer to pay for supported products as the cost will be passed on to the customer and they are willing to pay for quality. Do not want any complex scripting screwing around with third parties and such. Your ideas welcome. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to text for Asterisk
On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington ch...@acsdi.comwrote: All automated solutions -- paid or free -- are terrible. The technology simply does not exist at this point at a level that is acceptable to most customers. If quality is paramount, you are better off doing the transcription in-house with a human. In-house transcriptions are definitely out of the question, but any experience with outsourced solutions would be useful. As far as I can tell the current service is automated, and as awful as Google Voice, yet they find it useful. Their existing carrier uses Broadsoft and I'm not sure if they have that built in. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to text for Asterisk
On Mon, Oct 22, 2012 at 2:43 PM, Nickolay V. Shmyrev nshmy...@nexiwave.comwrote: There is no holy grail yet, speech technology deployment requires a close cooperation between the speech technology provider and the users. It's not plug and play but after some joint efforts automated transcriptions must be useful. If anyone wants to experiment with CMUSphinx-based automated solution to transcribe voicemails, drop me a note. The results could be pretty interesting. We will probably give it a try, though not sure when. Probably in about a month. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to post to list
On Tue, Oct 9, 2012 at 12:16 PM, Adnan 112linuxstockh...@gmail.com wrote: Hi who is responsible for this mailing list? i am not able to post to it. You just did. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On Thu, Oct 4, 2012 at 6:29 AM, Brett Lehrer brett.leh...@solarismed.comwrote: I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VoIP usage. For both incoming and outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful of reasons. Is it natural to have this many problems on a completely digital configuration? I'm trying to cut our analog phone line (because it's so expensive), but some fax machines just don't seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through. Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)? I'm not going to address the tech issues, as others already have. And if you didn't know, Steve Underwood is THE fax guy so whatever he says is gold, listen to him. However I'd just suggest that you look at the business case for screwing around with fax at all. As a society, if we had decided to stop supporting this dead technology years ago, with all the time and money we've collectively wasted we could have completely eliminated world hunger. I can't count the hundreds of hours I've wasted on fax support just to prop up this stupid and unnecessary technology. We just made the decision this week to outsource it all and never deal with it on our network again. I am slowly re-gaining my sanity because of that decision. Now I'm going to take a fax machine out to the parking lot and shoot it, even talking about this awful waste of time makes my blood boil. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On Thu, Oct 4, 2012 at 10:06 AM, Lee Howard fax...@howardsilvan.com wrote: I recognize that you're being a bit facetious in this latter comment No, not really. I stand by it. Useless and *should* be dead. It's dead and people just don't know it. There is no adequate replacement for fax. E-mail doesn't do it Yes, it does. Well, if you were using stand-alone fax machines then that was part of your problem. That was actually the only part of my post that was in jest. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. If you think 100 or 10k RTP ports going to your voice server makes ANY difference in security, you really need to re-think this and study more. Not to be a dick or anything, but really, think about it. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 8:38 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: I'm speaking of surface area. Ask any general if he would rather have to defend a 1000 mile front or a 1 mile front. You are right that an open port is an open port, but trying keeping the crowd out of 1 doors is *much* harder than trying to keep them out of 100 doors. Your trite comparison is irrelevant in this context. You are not protecting your 100 ports any more or less than 1000 or 10,000. But do as you will, I'll agree to disagree. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 8:46 AM, Eric Wieling ewiel...@nyigc.com wrote: A port is not a door if there is nothing listening on the port. Open ports are not a security issue. Stuff running on open ports are. In other words, a million ports with nothing listening is no worse than one with nothing listening. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users