Re: [asterisk-users] (no subject)
We have Cisco 9760 for executives and Aastra 9112i for everybody else. We started with Grandstream, don't remember the model, cost around $80 USD but it had bad audio quality and echo problems (running asterisk 1.09). The quality of construction felt poor, like a toy phone. We replaced them with the Aastra for double the cost and the quality improved dramatically. Audio quality was much better and echo problems all but eliminated. This phone also feels more solid. There are a few areas that are not perfect; the speaker phone is good not excellent and we have had to replace a couple of phones because they have stopped working. Over all I would say not bad for the price especially if they are for general use. We had to upgrade from the Aastra phones for our executives because they needed very good audio for both handset and speaker phone. We are using Cisco 9760's for them and have had no problems with quality. Plus they have a very solid feel. My question to the list is: As I need to add phones I am considering buying used Cisco 9760's. Is there any difference with the 9760G? I have heard that the 9761's have even better audio quality. Our main requirement is audio quality, our users do not need a lot of features on their phones. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 900 rules
810 is an area code in Michigan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Doug Crompton Sent: Tuesday, November 14, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 900 rules Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric ManxPower Wieling wrote: Doug Crompton wrote: I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? _1900NXX _NXX976 ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CAPI channel not available but nobody is usingthe system
Armin, I am running 1.2.7.1 with an Eicon T1 board version 2 on Debian 2.4 I don't know the details on chan-capi / CAPI drivers. We did the install April of this year. How can I tell what I have? Thank you for your time. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Armin Schindler Sent: Wednesday, October 18, 2006 3:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CAPI channel not available but nobody is usingthe system On Tue, 17 Oct 2006, Tim Sharp wrote: I have 23 CAPI channels defined and normally multiple channels are in use during the day for outbound calling. The problem is that every 3 or 4 months one of the channels becomes unavailable and then no calls can come in or go out on any of these channels. CAPI INFO shows Contr1: 23 B channels total, 22 B channels free. To fix the problem I reboot the asterisk server. First, is there a better way to reset the channels than rebooting? It depends where the problem really has its origin. If just asterisk (chan-capi) has a wrong channel count, it would be enough to unload chan-capi. Maybe asterisk itself need to be restarted. But if the real problem comes from the CAPI/ISDN driver, you need to reload these drivers. Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI driver do you use? Second, is there a way to bypass the unavailable channel in the dialplan? No. Third, what is causing the problem and can I prevent it? chan-capi counts the active channels when the CONNECT/DISCONNECT message of b-channels are indicated. If one of these messages are missing (it's a bug in the CAPI driver if that happens) the count is wrong. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAPI channel not available but nobody is using the system
I have 23 CAPI channels defined and normally multiple channels are in use during the day for outbound calling. The problem is that every 3 or 4 months one of the channels becomes unavailable and then no calls can come in or go out on any of these channels. CAPI INFO shows Contr1: 23 B channels total, 22 B channels free. To fix the problem I reboot the asterisk server. First, is there a better way to reset the channels than rebooting? Second, is there a way to bypass the unavailable channel in the dialplan? Third, what is causing the problem and can I prevent it? Thank you in advance. Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk upgrade
I am currently running 1.2.7.1 and it works just fine. I personally like to stay 3 or 4 months behind the current release. This time it is a bit longer because I don't feel comfortable with the stability of later releases. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Conrad Wood Sent: Monday, October 16, 2006 7:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk upgrade On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote: at the moment (fortunately) i'm not experiencing any kind of particular problem, do you suggest me to upgrade asterisk? #1 sysadmin rule: If it's not broken, just don't fix it. That will get you into trouble when it _does_ break. I rather *test* new versions, fix any configuration problems and then keep the live versions uptodate. It can be quite a nightmare to skip lots of versions, particularly under timepressure with a broken system at hand. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hardware ? Analog DID trunks (ILT)
I am looking at CTPX's VP2000 product. I haven't tried it yet. Please let me know if you find a solution that works. Tim -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jonn R TaylorSent: Friday, September 01, 2006 12:15 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Hardware ? Analog DID trunks (ILT) Is there a card that supports analog DID trunks, alosi known as ILT trunks or Incoming Loop Trunk. They work by providing talk battery to the CO, incoming calls happen by pulling loop sending a wink accepting the DID dtmf digits for the station being called. Jonn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anybody using Eicon SoftIP with Asterisk
I am trying to configure Eicon's SIP version of SoftIP to work with Asterisk. The documentation is poor and my general knowledge of SIP communications is limited. I am getting a circuits busy message when trying to call the IP address of the server with SoftIP. If anybody has gotten this to work please respond. Thank you. Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error: Dropping incompatible voice frame
Hello, I get this error message when trying to route an incoming fax from a packet based T1 to an EICON board that is connected to an external fax voice mail server. Voice calls route to this external server with no error. Both fax and voice calls that come in a channelized T1 also route to this external server with no errors. I am on 1.2.7.1 Jul 13 13:19:56 NOTICE[24867]: channel.c:1904 ast_read: Dropping incompatible voice frame on CAPI/PRI1/XX-14a of format slin since our native format has changed to ulaw I did find a reference to something similar in Mantis issue tracker number 0004101. I am not technical enough to know if this is the same issue. Any help to explain what is the problem or how to fix it is greatly appreciated. Thank you, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Voice calls sent to fax extension
I also had an intermittent problem, on average one or two faxes a week, that were not recongnzed as a fax. Then I switched phone companies and have not had that problem since. It has been over 2 months. I addition, my echo problem has been practically eliminated and overall voice quality is better. I believe that it is because of better levels of RX gain for fax recognition, but don't know for sure. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul A. Pringle Sent: Monday, June 26, 2006 10:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Voice calls sent to fax extension I thought it might be an inadvertent button press, but none of the keys (on my phone at least) are recognized by Asterisk as fax tones. This has happened to two different users getting calls from different people using different equipment. Does anyone else see this behavior occasionally? Paul -Original Message- Date: Fri, 23 Jun 2006 15:29:31 -0400 From: Bill Gibbs [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice calls sent to fax extension To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SE Michigan asterisk users group
Steven, I am in Livonia. Please let me know if there is interest. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BerkHolz, Steven Sent: Thursday, June 22, 2006 4:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SE Michigan asterisk users group I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven http://www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asking for phone number to dial
The number dialed after Background is stored in the EXTEN variable and can be used in the Dial application. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Don Sent: Friday, June 23, 2006 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asking for phone number to dial background just accepts input while other sounds...etc...are being played... instead of waiting for something to end and then accept input. It doesn't store the number...etc...then add it to dial command for a zap channel - Original Message - From: T. Shaw [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 23, 2006 5:19 PM Subject: RE: [Asterisk-Users] Asking for phone number to dial Isn't that what the Background() application does? [EMAIL PROTECTED] blah... From: Don [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asking for phone number to dial Date: Fri, 23 Jun 2006 15:51:00 -0400 Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use them in the dial command... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.2/373 - Release Date: 6/22/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing?
The options are not seperated by commas. exten = s,1,Dial(SIP/50,23,r,d) should be exten = s,1,Dial(SIP/50,23,rd) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing? Any idea why it wouldn't work in my dial plan? On 6/22/06, Peter Antonacci [EMAIL PROTECTED] wrote: d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also On 6/22/06, John Klimek [EMAIL PROTECTED] wrote: Anybody have any more information on this Dial() d option for incoming calls? On 6/19/06, John Klimek [EMAIL PROTECTED] wrote: Thanks for the information... After doing some reading it looks like I can use the d option with the Dial() command to be able to enter a 1-digit extension while the other extension is ringing, but this doesn't seem to be working for me either... Here is my new config: exten = s,1,Dial(SIP/50,23,r,d) exten = s,2,VoiceMail( [EMAIL PROTECTED]) exten = s,3,Playback(vm-goodbye) exten = s,4,Hangup exten = 1,1,SayDigits(1) exten = 2,1,SayDigits(2) exten = 10,1,SayDigits(10) However, when my phone is ringing (eg. extension 50), I try entering 1 or 2 (to be forwarded via the Dial d option), but it doesn't do anything. What am I doing wrong? I like your solution above, but if I use that I'll need to wait 23 seconds for Dial() to timeout before I can do anything. I'd like to be immediately able to enter an extension (if possible, which maybe it's not...) On 6/19/06, Leah Newmark [EMAIL PROTECTED] wrote: Using the Background command, you will be able to play the voicemail while still being allowed to enter digits. exten = s,1,Wait(2) exten = 108,2,Background(voicemail/default/108/unavail) exten = s,1,Dial(SIP/50,23,r) exten = s,2,Background(/voicemail/default/50/unavail) ;or whatever the soundfile is called exten = s,3,Voicemail(s50) ;s will skip the greeting and just go to the beep exten = s,4,Playback(vm-goodbye) exten = s,5,Hangup You can then put exten = 1, Dial(sip/me) exten = 2, Dial(sip/her) or whatever your dial statements look like. Leah Newmark Capalon VoIP [EMAIL PROTECTED] wrote: Message: 9 Date: Mon, 19 Jun 2006 14:18:22 -0400 From: John Klimek [EMAIL PROTECTED] Subject: [Asterisk-Users] Can I enter an extension to dial while voicemail is playing? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten = s,1,Dial(SIP/50,23,r) exten = s,2,VoiceMail([EMAIL PROTECTED]) exten = s,3,Playback(vm-goodbye) exten = s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are forwarded to my voicemail. How can I make it so I can call an enter extensions either while the phone is ringing or while the voicemail message is playing? I want the system to be as seemless as possible so the wife is happy =) Right now it works great because my Sipura 3000 forwards to call to Asterisk and Asterisk rings my analog phone, but the incoming caller hears a steady dial-tone the whole time. I wouldn't want that to change. (so the caller isn't wondering what is going on) Any help is appriciated :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
[Asterisk-Users] No ring tone on outgoing calls
I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No ring tone on outgoing calls
Here is a copy of indications.conf I have not included other country codes. I noticed that there are no spaces in country=us does that matter? Thanks [general] country=us [us] description = United States / North America ringcadance = 2000,4000 dial = 350+440 busy = 480+620/500,0/500 ring = 440+480/2000,0/4000 congestion = 480+620/250,0/250 callwaiting = 440/300,0/1 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric ManxPower Wieling Sent: Wednesday, June 14, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No ring tone on outgoing calls Make sure you have /etc/asterisk/indications.conf set up. People that don't know any better might tell you to use the r option to Dial. Those people are confused. Don't do that until you have tried everything else. Tim Sharp wrote: I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback welcome message while phones ring, please help
Yes it does, I just set our system up that way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gareth Blades Sent: Tuesday, June 06, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback welcome message while phones ring,please help I believe if you use the new native music on hold feature it always plays the music on hold starting from the beginning. On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote: I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback welcome message while phones ring, please help
I am running on 1.2.7.1 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of OlivierSent: Tuesday, June 06, 2006 12:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Playback welcome message while phones ring,please help 2006/6/6, Gareth Blades [EMAIL PROTECTED]: I believe if you use the new native music on hold feature it alwaysplays the music on hold starting from the beginning.Where can I find this "new native music on hold feature" ?In Asterisk 1.2.x ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on Hold restart at beginning for each call
FYI, we upgraded to 1.2.7.1 this weekend and the new features in MOH allow for the sound file to start at the begining with each call. Thanks to Chris and Matt for their responses. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tim Sharp Sent: Friday, May 19, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music on Hold restart at beginning for each call Chris, The queues idea is a good one. I will check it out. Thanks for all of your suggestions. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Hastie Sent: Friday, May 19, 2006 3:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for each call On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote Chris, When I tried background it waited until the message was done before dialing, just like playback. Am I missing something? Wasn't my suggestion :) If I've understood what you're trying to I would go one of two ways: Rather than dial each of the four numbers sequentially, dial them simultaneously. This should hopefully speed up your average pick up time, but will loose any control over preference for who deals with the call. Or investigate queues. I don't have enough people to make it worth my while looking at these, so I've no idea if they're what you need, but they sound like they might be. -- Chris Hastie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on Hold restart at beginning for each call
Chris, The queues idea is a good one. I will check it out. Thanks for all of your suggestions. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Hastie Sent: Friday, May 19, 2006 3:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for each call On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote Chris, When I tried background it waited until the message was done before dialing, just like playback. Am I missing something? Wasn't my suggestion :) If I've understood what you're trying to I would go one of two ways: Rather than dial each of the four numbers sequentially, dial them simultaneously. This should hopefully speed up your average pick up time, but will loose any control over preference for who deals with the call. Or investigate queues. I don't have enough people to make it worth my while looking at these, so I've no idea if they're what you need, but they sound like they might be. -- Chris Hastie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on Hold restart at beginning for each call
Chris,
When I tried background it waited until the message was done before dialing,
just like playback. Am I missing something?
Thanks,
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Sent: Monday, May 15, 2006 4:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for
each call
Why not use background?
On 5/15/06, Tim Sharp [EMAIL PROTECTED] wrote:
Chris,
I have it working now using Playback but I am looking for a way to reduce the
wait time.
The system can hold up to four numbers per each DID number. When a call
comes in the first message says
Please wait while I attempt to connect you. At any time during this call
you may press 1 to go straight
to voice mail. And the first of four numbers is dialed. If the call is not
answered then the second
message says Thank you for your patience. I am trying an alternate number.
Remember you can always
press 1 to go staight to voice mail. And the next number is dialed. This
can continue for two more
numbers. If there is no answer and all numbers have been tried the message
I am sorry I was unable to
locate your party. I will transfer you to their voice mail. And the system
transfers the call to voice mail.
The best case would be to dial the number while playing the message. I have
this working except that the
sound file is played in a loop when using the m option with dial. The first
call works great, every call after
the message will start where the prior call ended. I am looking for a way to
have the sound file start at the
begining with each call.
Thanks,
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Hastie
Sent: Sunday, May 14, 2006 5:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for
each call
On Fri, 12 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
I am using the m option on the dial command to play a message instead
of ringing. The message is something like please wait while I try to
locate your party so I need it to start at the beginning for each
call. I think there might be a way in 1.2.x be we are not ready to
upgrade yet so a solution for 1.0.9 is what I am after. Thanks.
The 'm' option is for music on hold, not really announcements. Wouldn't
it be simpler to play the announcement first, then dial? eg
exten = s,1,Playback(local/please_hold_locate)
exten = s,2,Dial(${FOO},20,tm)
--
Chris Hastie
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on Hold restart at beginning for each call
Chris,
I have it working now using Playback but I am looking for a way to reduce the
wait time.
The system can hold up to four numbers per each DID number. When a call comes
in the first message says
Please wait while I attempt to connect you. At any time during this call you
may press 1 to go straight
to voice mail. And the first of four numbers is dialed. If the call is not
answered then the second
message says Thank you for your patience. I am trying an alternate number.
Remember you can always
press 1 to go staight to voice mail. And the next number is dialed. This can
continue for two more
numbers. If there is no answer and all numbers have been tried the message I
am sorry I was unable to
locate your party. I will transfer you to their voice mail. And the system
transfers the call to voice mail.
The best case would be to dial the number while playing the message. I have
this working except that the
sound file is played in a loop when using the m option with dial. The first
call works great, every call after
the message will start where the prior call ended. I am looking for a way to
have the sound file start at the
begining with each call.
Thanks,
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Hastie
Sent: Sunday, May 14, 2006 5:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for
each call
On Fri, 12 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
I am using the m option on the dial command to play a message instead
of ringing. The message is something like please wait while I try to
locate your party so I need it to start at the beginning for each
call. I think there might be a way in 1.2.x be we are not ready to
upgrade yet so a solution for 1.0.9 is what I am after. Thanks.
The 'm' option is for music on hold, not really announcements. Wouldn't
it be simpler to play the announcement first, then dial? eg
exten = s,1,Playback(local/please_hold_locate)
exten = s,2,Dial(${FOO},20,tm)
--
Chris Hastie
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold restart at beginning for each call
I am using the m option on the dial command to play a message instead of ringing. The message is something like please wait while I try to locate your party so I need it to start at the beginning for each call. I think there might be a way in 1.2.x be we are not ready to upgrade yet so a solution for 1.0.9 is what I am after. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
