Re: [asterisk-users] Testing 911 call
I actually work in a 911 center. Please do not dial blindly to do a test call. Please call the non-emergency dispatch number, ask if it would be ok to make one or two test calls. If they give you the ok, please complete those calls as quickly as possible as conditions change in an instant. If they give you permission at 9am, don't wait until 5pm to do the test. Further, most 911/PSAP centers are not busy after 10pm local time, please consider doing your testing in the overnight hours. Again, please check with your local 911/PSAP to confirm when their peak times are and try to avoid them. Here is a good script to read when speaking to the 911 call taker: Hello my name is (your name) with (company name). We are performing a test 911 call and would like to confirm some information if you have a moment. (if they answer go ahead, continue with the script. If they advise now is not a good time, say thank you and disconnect. In a 30 to 60 minutes call the non-emergency number and ask if you may make another test call) (continuing the script) Can you please confirm the address that shows up on your phone system please? (wait and confirm the info) Great thank you. If you can, please tell me the number you show we are calling from? (wait and confirm) Can you confirm for me which 911/PSAP center we have reached? ( wait and confirm this is the proper 911/PSAP you need) (if this is the first of several calls:) Thank you very much for your time, we will be making (xx number) of calls in the next few minutes. (if this is the end of your testing:) Thank you very much for your time, this concludes our testing. Good luck with your phone testing. Regards, James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 On Sun, May 5, 2013 at 8:00 PM, Dale Noll wrote: > If there is a non-emergency number you can call and let them know you > would like to do some test calls. This also allows you to schedule a time > for testing when the PSAP is not as busy allowing for real calls to be > handled. > > > On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt < > ma...@intuitiveengineering.com> wrote: > >> Joseph, >> >> I have made a quite a few test calls to 911. They don't charge you and >> they don't get upset. >> >> Just let them know right away it is a non-emergency test call, and then >> let them know who you are and what you need to verify on their information >> screen. >> >> Mark Engelhardt >> >> >> On May 5, 2013, at 11:07 AM, Joseph wrote: >> >> > How to test 911 call? >> > >> > I'm using Audiocodes and it setup to strip the first number but I've >> never tested the 911 call. I don't want to go live as they might charge me. >> > >> > -- >> > 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-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] Reliable SIP Trunk Provider
You might check flowroute. We have been with them for over a year now and have been spot on with service and support. www.flowroute.com and they are one of the cheapest providers we have found for our needs. Regards, James Miller Agent Black Web Hosting "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 On Thu, Mar 15, 2012 at 11:45, Jake Wicke wrote: > I'm wondering if any other Asterisk users have a recommendation for a > reliable SIP Trunk provider that supports Asterisk and offers decent > support. > > I've worked with Coredial, Broadvox, and Broadvoice and have had some bad > experiences with each of these providers. > > Broadvoice offers low cost service, however I have constant issues with > Broadvoice blocking my customers due to Asterisk "registering too often". > Support either does not respond to e-mails, hangs up on phone calls, or > gives me the "we don't support Asterisk and we can use your account no > problem using the SIP phone on our desk" line. > > Coredial resigned me into a two year agreement after making a change to my > SIP trunk configuration without my knowledge, then demanded two years of > the full monthly charge when I tried to cancel over a dispute regarding > services that I did not order. Check out coredialhorrorstory.com for the > whole story. While the service is decent, the customer service leaves much > to be desired. > > Broadvox has been the best provider that I have found so far, however I > initially had a lot of issues with sales quoting a product which could not > be provisioned and also not being able to deliver service on a timely > schedule. I also was given the run around by customer service recently on > a simple request to add a DID number to an account. > > Thanks for your input! > > > > -- > _ > -- 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] Cisco 7940 phone and tftpd provisioning - for ever?
I have my tftp daemon running all the time and it really doesnt affect the performance of the machine. Is there a reason why you want to shut it down? “I see blindness more as the ability and sight more as the disability, I only see that which is within a person.” Patrick Henry Hughes - 2009 On 5/8/2011 5:19 PM, Sebastian Arcus wrote: Hi all, Sorry for posting here - but I figured there are many people with Cisco IP phones here - and I use them with Asterisk :-) I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK, loaded the SIP configuration files OK, they work with Asterisk just fine. My question is - will I have to keep on running the tftp server for them for ever and ever? Isn't there any option for them to just use the settings they have already loaded form the tftp server - so that I can kill tftpd on my server machine? I tried doing that, and then the phones stop booting, going in a loop looking for the tftpd server. It seems a bit pointless, having to run the tftpd daemon all the time - although I've already loaded the firmware and configurations I want. Thank you, Sebastian -- _ -- 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] Cisco 7960 & asterisk 1.8.22 ringlist.dat error
Problem has been resolved with the assistance of Jonathan. Appears to be an issue with my text editors not properly tabbing the file correctly. Regards. "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. -- _ -- 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 7960 & asterisk 1.8.22 ringlist.dat error
Yes, nothing changed EXCEPT for the software image the phone pulled down. All of the files are still in the exact same locations with the exact same names as they had in 8.9. I'm at a loss as to what's causing this issue and so apparently is Cisco given they have yet to respond to my follow up information. Regards. "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. Do you actually have those files in your TFTP directory? You need both the RINGLIST.DAT file that specifies what files are available and what they are called, PLUS the actual ring files themselves. All of my Cisco ringer files are .pcm files, like "ATT,pcm", "ATT2.pcm", etc. Tom -- _ -- 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 7960 & asterisk 1.8.22 ringlist.dat error
I did that and this is what I got when I tried to play the 24 ringtone: 13:29:16.573318 IP 192.168.1.103.50849 > 192.168.1.60.69: 39 RRQ "Emergency ring_emergency.pcm" octet In the ringlist.dat file in the first column I typed the display name then hit the tab key. Now on some it only moved a couple of spaces over, on others, it tabbed way over. Not sure whats going on there with that. Thank you for your help. "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. -Original Message- > I did the command listed, and its actually requesting RINGLIST.DAT, so I > changed the filename to match its request but now its showing in the ring > type setting: > > Chirp 1 > Chirp 2 > 24 24-ring-tone-1.raw > Att1 ring_att1.pcm > . You should only see the description of the file on the display. Make sure that the description and filename are tab-separated, since spaces are allowed in the description. > However, when you attempt to play one it says Loading Ringer File but it > doesnt do anything. So now its at least seeing the file, now it just > wont play them. You can run the same command ( tcpdump -nn port 69 ) to view what file the phone is attempting to download from the tftp server. My guess is that it isn't pulling anything down or something like "24 24-ring-tone-1.raw" if the file is not tab separated. -Jonathan -- _ -- 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 7960 & asterisk 1.8.22 ringlist.dat error
I did the command listed, and its actually requesting RINGLIST.DAT, so I changed the filename to match its request but now its showing in the ring type setting: Chirp 1 Chirp 2 24 24-ring-tone-1.raw Att1 ring_att1.pcm . . . However, when you attempt to play one it says Loading Ringer File but it doesn't do anything. So now it's at least seeing the file, now it just won't play them. Thanks for the help thus far! James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Monday, February 14, 2011 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error On Mon, Feb 14, 2011 at 5:40 AM, James Miller wrote: Error! Filename not specified. Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. Have you run a tcpdump on the tftp server to make sure it is requesting the correct file? It might be asking for RingList.dat, ringlist.dat, RINGLIST.DAT, etc. as capitalization seems to not be one of Cisco's concerns. (FYI, mine was RINGLIST.DAT, but I have no more 79x0's around to test with) Try running this as root on the tftp server and look for a request for the file: # tcpdump -nn port 69 -Jonathan -- _ -- 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 7960 & asterisk 1.8.22 ringlist.dat error
That's the problem, I am not sure if the problem lies with Cisco, or if it lies with Asterisk. I figured I'd try here first before running in circles with a TAC Case. Regards. "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif Sent: Monday, February 14, 2011 8:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error Sensitivity: Confidential Better to report a BUG to cisco. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller Sent: Monday, February 14, 2011 6:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error Sensitivity: Confidential Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. I have asterisk 1.8.2.2 installed with the FreePBX module with a 7960 just recently flashed to 8.12. Not sure what else you all may need but any help would be greatly appreciated. Respectfully, James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. <>-- _ -- 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 7960 & asterisk 1.8.22 ringlist.dat error
Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. I have asterisk 1.8.2.2 installed with the FreePBX module with a 7960 just recently flashed to 8.12. Not sure what else you all may need but any help would be greatly appreciated. Respectfully, James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. <>-- _ -- 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
When you get over 500 emails a day on your blackberry you have make a decision on what is or is not worth reading at that moment. Its not lazy at all its cutting through the fluff and finding the emails worth while. When inside outlook you don't have the hot key b to scroll to the bottom so again, I'd have to scroll down. Add up the time it takes per email x 500 emails, you loose considerable amount of productivity. Top posting has its useful place as well as bottom posting. Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Fred Posner Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 21:43:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Mon, 2011-01-17 at 02:31 +0000, James Miller wrote: > I hate to disagree but I find it much, much easier to follow conversations > when the newest reply is on top. I find it too time consuming to scroll > through a long message just to find out someone left a three word reply. > > As I am on my blackberry more than I am at a pc, if I don't see the reply as > soon as I open the message it gets deleted without being read. Time is money > and I don't have time to scroll through every message. > > I will agree that sometimes it is helpful to make replies at the bottom and I > will attempt to keep the peace by posting at the bottom when I can, but top > posting is easier and more clean to read than having 100 lines of > and > broken lines. > > Warmest regards, > James > > > Sent from my Verizon BlackBerry. Always on, Always Connected > > -Original Message- > From: Lesly Dorval > Sender: asterisk-users-boun...@lists.digium.com > Date: Mon, 17 Jan 2011 02:14:54 > To: > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Top Posting > > Shaun Ruffell digium.com> writes: > > > > > Whatever your preferred style, the following post is at least worth > > considering. > > > > http://brooksreview.net/2011/01/interleaved-email/ > > > > My belief is that it would be nearly impossible for me to follow a high > > volume list if top posting was the preferred style. For example, the > > following email from the LKML would need to be more verbose if all the > > participants were top posting, because they would all have to set the > > context for their comments. Instead, you can follow the chain of > > thought for each of the "threads" contained in the email. > > > > http://article.gmane.org/gmane.linux.kernel/1087665 > > > > Anyway, just something to consider, > > Shaun > I could never understand the strong objection regarding top-posting until > Shaun > shared these examples - though I had been reading lists for more years than I > care to admit. These examples clearly show how snipping and bottom posting > translate to susccint and clear contextual communication. From now I will > evangelize snipping and bottom posting. > I cannot imagine considering scrolling to the end of an email time consuming. Very sad. If you find it too difficult on your blackberry to press the B key (to jump to the bottom of the message) then I am uncertain how you have enough time to even read this email. I'm all for good arguments. That "time consuming" one is just lazy. I personally find top posting annoying and only serving to an immediate conversation. Particularly useless if referencing the message later. -- With best regards, ---fred http://qxork.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
Re: [asterisk-users] Top Posting
I hate to disagree but I find it much, much easier to follow conversations when the newest reply is on top. I find it too time consuming to scroll through a long message just to find out someone left a three word reply. As I am on my blackberry more than I am at a pc, if I don't see the reply as soon as I open the message it gets deleted without being read. Time is money and I don't have time to scroll through every message. I will agree that sometimes it is helpful to make replies at the bottom and I will attempt to keep the peace by posting at the bottom when I can, but top posting is easier and more clean to read than having 100 lines of > and broken lines. Warmest regards, James Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Lesly Dorval Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 17 Jan 2011 02:14:54 To: Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting Shaun Ruffell digium.com> writes: > > Whatever your preferred style, the following post is at least worth > considering. > > http://brooksreview.net/2011/01/interleaved-email/ > > My belief is that it would be nearly impossible for me to follow a high > volume list if top posting was the preferred style. For example, the > following email from the LKML would need to be more verbose if all the > participants were top posting, because they would all have to set the > context for their comments. Instead, you can follow the chain of > thought for each of the "threads" contained in the email. > > http://article.gmane.org/gmane.linux.kernel/1087665 > > Anyway, just something to consider, > Shaun I could never understand the strong objection regarding top-posting until Shaun shared these examples - though I had been reading lists for more years than I care to admit. These examples clearly show how snipping and bottom posting translate to susccint and clear contextual communication. From now I will evangelize snipping and bottom posting. -- _ -- 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] Music on Hold not working?
That's what I'm wanting to change. I want it to stream 100% of the time no matter if the person is in queue or if an agent has answered. Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Warren Selby Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 12:41:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? MOH plays the default class unless specified by a channel variable to play a different one. In queues.conf you can specify the MOH class on a queue by queue basis, but that's the hold music for someone waiting to be answered. Once an agent answers, if they put someone on hold they'll be put into the default MOH class unless a channel variable is specified beforehand. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 11:55 AM, "James M Miller" wrote: > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen > Sent: Sunday, 16 January, 2011 12:47 > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Music on Hold not working? > > On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: >> I'm going out on a limb here, as I'm still pretty new to Astrisk and > running >> my own VOIP server, however I believe there is a bug or flaw with the > Music >> on Hold feature. >> >> I have it all configured and it should work, and it did briefly several >> weeks ago, however now, it doesn't work at all and only plays the default >> hold music. > > Well, why should it play something different? > > How have you configured it to play something different? > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > -- > _ > -- 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 > [James M Miller] > > Well one would think that if you configure the Music on Hold feature by > setting streams for it to pull from, it should play it no matter how the > phone is dialed. > > Meaning if I dial another extension on the network, I should hear the MOH > since I have it programmed with streams. However what is occurring is it is > only playing when you are placed into a queue. Once someone picks up the > line, it starts playing the default again if that person places the person > on hold. > > One would think that it would play MOH no matter what if you have the > streams programmed and override the defaults, at least that's what I'd like > for it to do. > > Regards, > James > > > -- > _ > -- 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] Music on Hold not working?
>From the command you suggested to enter: Class: default File: /var/lib/asterisk/moh//reno_project-system File: /var/lib/asterisk/moh//macroform-robot_dity File: /var/lib/asterisk/moh//manolo_camp-morning_coffee File: /var/lib/asterisk/moh//macroform-cold_day File: /var/lib/asterisk/moh//macroform-the_simplicity Class: none File: /var/lib/asterisk/moh/.nomusic_reserved/silence Basically the queues will stream the online music, but if I call another extension on the network, it will play just the default sounds. One would think that if you have suggested the system play streaming music for everything else, it would follow suite and play streaming for ext to ext calls. "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Allen Sent: Saturday, January 15, 2011 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 9:36 AM, James Miller wrote: Forgive me, but how do I do moh show files? Basically what is occurring is: If you enter a queue and are waiting to be answered, you will hear the streaming MOH If you call another extension on the system, you will only hear the default MOH. I want it to stream MOH for everything. Hopefully that makes sense. Regards, James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Allen Sent: Saturday, January 15, 2011 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from "moh show files" in Asterisk? Go into Asterisk CLI (asterisk -r) and issue the command "moh show files". I don't see how you can have different MOH in a queue vs. being on hold unless you have specified a specific MOH group for your call queues. -- _ -- 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] Music on Hold not working?
I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. I have tried noload => res_timing_dahdi.so in the etc/asterisk/modules.conf file, however that doesn't work. I've tried moving that line to the very top, very bottom, and several places in the middle, and still can not get MOH to work. I don't use dahdi, nor do I use the local Telco lines. It is strictly a VOIP trunk. Yes I know there are advantages and disadvantages to this kind of set up, however, I have a failover route set up and configured with the trunk provider, so I am ok with what I have at this point. Currently AstriskNOW/FreePBX is installed on: Dell precision 490 workstation Dual xeon 5100 dual core processors 4gb ECC Buffered Ram 73GB 15krpm hard drive The Machine has Xen server installed with asterisknow running on its own VM with: 2 cores 2gb ram 40gb hard drive space I installed AsteriskNow 64bit image that is available for download from Freepbx's website, and have done all of the updates that both the freepbx interface, as well as CLI yum update command suggests with no fixing of the problem. I thank you all in advance for taking the time to read this issue and look forward to hopefully fixing my MOH. Warmest Regards, James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. -- _ -- 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 Asterix right tool for me?
The simple answer: it takes the digital (VOIP) signals, and connects your calls to the traditional landline network. So if you are in Chicago, and want to call New York City, it will take your call from your office in chicago, route it, and terminate the call to the regular LandLine provider to talk to Joe's Deli in New York City. And the reverse happens. Someone calls your VOIP number, it connects to your SIP Trunk provider, converts it to digital, and sends it to your VOIP system. If i misspoke something, someone please feel free to correct me, but this is my understanding of it. The technical answer, I will have to defer you to someone with more technical knowledge than myself. For example, my SIP Trunk is provided by www.flowroute.com . They seem good thus far. They came at the suggestion of a friend who uses them. They are a "prepaid" service. You have to put money into an account, and the monthly charges and the cost per call are charged towards that balance. I am in the process of building out FreePBX which uses asterisk for my web hosting business. So far it has worked well. I've tested the call quality on Flowroute and it has been good. Several people i have talked to cant tell im using VOIP. Shop around, find a provider that works for you, and ask for opinions. Good luck, and i hope i made it as clear as mud! :) James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:48, James Miller wrote: > Thats right, i completely forgot that option! I run a soft phone on my > laptop, which connects back through my verizon wireless aircard to the pbx > and allows me to call out while on the go from anywhere! > > > "I see blindness, not as a disability, but more of an ability. And Sight > actually, more of a disability because some people with sight tend to judge > others by what they see on the outside, whereas I don't see that. I just see > that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 > > > On Wed, Oct 20, 2010 at 03:36, GBR Icasiano, Ryan A. < > raicasi...@globalbridgeresources.com> wrote: > >> i think you can also use softphones installed in your remote offices. >> >> regards, >> >> RYAN ICASIANO >> ________ >> From: asterisk-users-boun...@lists.digium.com [ >> asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller [ >> paramedi...@gmail.com] >> Sent: Wednesday, October 20, 2010 3:34 PM >> To: exp...@hope.cz; Asterisk Users Mailing List - Non-Commercial >> Discussion >> Subject: Re: [asterisk-users] Is Asterix right tool for me? >> >> In short terms: >> >> 1)broadband internet connection >> 2) Voip phone like a Cisco 7960 >> 3) Sip Trunks from a SIP Trunk provider >> >> Thats a short list of what you will need, but you could ditch your local >> Telcom operator completely, and run VOIP. >> >> There are much more knowledgable people about the subject matter than me, >> but this should at least get you started! >> >> Good luck and Welcome to Asterisk! >> >> James >> >> >> "I see blindness, not as a disability, but more of an ability. And Sight >> actually, more of a disability because some people with sight tend to judge >> others by what they see on the outside, whereas I don't see that. I just see >> that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 >> >> >> On Wed, Oct 20, 2010 at 03:22, > jana1...@centrum.cz>> wrote: >> Hi , >> I am a newbie with Asterix and not sure if Asterix is a right tool for my >> needs. >> >> Let's suppose this scenario : >> I have a telephone line in one office( all calls are paid to telephone >> operator). >> In other offices I have only internet connections. >> Is it possible to use Asterix so that I can make telephone calls from ALL >> offices( without >> direct telecom connection) ? if so, what telephone equipment would they >> have to use (VoIP >> telephones?) >> >> Thanks >> Jane >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com< >> http://www.api-digital.com/> -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.aster
Re: [asterisk-users] Is Asterix right tool for me?
Thats right, i completely forgot that option! I run a soft phone on my laptop, which connects back through my verizon wireless aircard to the pbx and allows me to call out while on the go from anywhere! "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:36, GBR Icasiano, Ryan A. < raicasi...@globalbridgeresources.com> wrote: > i think you can also use softphones installed in your remote offices. > > regards, > > RYAN ICASIANO > > From: asterisk-users-boun...@lists.digium.com [ > asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller [ > paramedi...@gmail.com] > Sent: Wednesday, October 20, 2010 3:34 PM > To: exp...@hope.cz; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] Is Asterix right tool for me? > > In short terms: > > 1)broadband internet connection > 2) Voip phone like a Cisco 7960 > 3) Sip Trunks from a SIP Trunk provider > > Thats a short list of what you will need, but you could ditch your local > Telcom operator completely, and run VOIP. > > There are much more knowledgable people about the subject matter than me, > but this should at least get you started! > > Good luck and Welcome to Asterisk! > > James > > > "I see blindness, not as a disability, but more of an ability. And Sight > actually, more of a disability because some people with sight tend to judge > others by what they see on the outside, whereas I don't see that. I just see > that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 > > > On Wed, Oct 20, 2010 at 03:22, jana1...@centrum.cz>> wrote: > Hi , > I am a newbie with Asterix and not sure if Asterix is a right tool for my > needs. > > Let's suppose this scenario : > I have a telephone line in one office( all calls are paid to telephone > operator). > In other offices I have only internet connections. > Is it possible to use Asterix so that I can make telephone calls from ALL > offices( without > direct telecom connection) ? if so, what telephone equipment would they > have to use (VoIP > telephones?) > > Thanks > Jane > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com< > 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] Is Asterix right tool for me?
In short terms: 1)broadband internet connection 2) Voip phone like a Cisco 7960 3) Sip Trunks from a SIP Trunk provider Thats a short list of what you will need, but you could ditch your local Telcom operator completely, and run VOIP. There are much more knowledgable people about the subject matter than me, but this should at least get you started! Good luck and Welcome to Asterisk! James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:22, wrote: > Hi , > I am a newbie with Asterix and not sure if Asterix is a right tool for my > needs. > > Let's suppose this scenario : > I have a telephone line in one office( all calls are paid to telephone > operator). > In other offices I have only internet connections. > Is it possible to use Asterix so that I can make telephone calls from ALL > offices( without > direct telecom connection) ? if so, what telephone equipment would they > have to use (VoIP > telephones?) > > Thanks > Jane > > > -- > _ > -- 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] Cisco SIP 8.5 and 9.0 Issues
I know this doesn't answer your question directly, but Where are you getting the Sip 9.0 software? It is not available on Cisco's website. I have Sip 8.9 on my phone and I have noticed that after about 45 mins on a call it will hang up and drop the desktop connection that runs through the phone. I am hoping that upgrading to 8.12 will fix the issue, or I just wasted money on a SMARTnet contract. Regards, James "I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person." Patrick Henry Hughes, Louisville Kentucky,2008 On Tue, Oct 5, 2010 at 17:08, Gerard wrote: > Hi list, > I was wondering if anyone had any solution to either one of two issues > I'm having: > I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, > it works very well for the most part, but after less then a week of > heavy usage, eventually the phone gets into a state where it won't > accept or let you place any more calls, the screen flashes "no free > lines available" or something along those lines. (power cycle fixes this). > So my preferred solution would be to upgrade to the v9.0(3) firmware, > but when that's loaded, the phone won't register with Asterisk anymore, > does anyone know if I need to adjust my .cnf.xml file, or is it a bug of > some sort? > Thanks for any input, > -- > Gerard Saraber > Network Admin. > Rarcoa, Inc > > -- > _ > -- 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
