Re: [Asterisk-Users] X100P blows up after a while (really loud noise)
Marconi, I don't know if this is will help you, but I had problems with some TDM400p cards. They worked fine, but after about 10 minutes in use there was a very loud static, humming noise. The cards where brand new, rev. G. I spoke with Digium about the problem, and they suggested that I update to the latest Asterisk, as there was a driver change in the last month (I was running a version from July). So I updated Asterisk, rebooted, and now my cards work great. Hope that helps! -Tor Marconi Rivello wrote: Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server, restarted the Asterisk (just the software). Didn't work. So I stopped it, unloaded the wcfxo module, loaded it up again and it was just fine. I could call the FXO and use it just fine. Weird. Last night, I talked for about 2 hours straight. No problem. But, this morning, when someone called the FXO all that could be heard was that loud noise. I could make a "stop-asterisk; reload modules; start-asterisk" script, and a cron entry or something to do it periodically, even check to see if there's any call on progress before restarting, but that's just a very ugly solution... If I could check the wcfxo status and get some info that tells me if it's in "buzzer-mode-on", I could come up with a more elegant solution. I don't know if it helps: the FXO card, at the first day, was sharing IRQ with the soundcard. But there wasn't any software using the soundcard. Yesterday, I unloaded all the sound modules, and checked /proc/interrupts. No IRQ sharing... But the problem occurred again later... In this cheap MoBo there's no option to mess around with IRQs in the BIOS. Today, I'm gonna disable onboard sound, to see if it helps at all, but I think that without modules loaded, it would have the same effect. I'll try this just to make sure... Did someone have this problem too? Any ideas, thoughts, suggestions...? Thanks, Marconi Rivello. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
John, Below are the impotant parts of my oss.conf and extensions.conf that make the system work. The only problems that I had in getting this to work was finding the right soundcard. I tried serveral of them, some just work and others don't. Be prepared to try a few out. -Tor Roberts This is the oss.conf. I think it is the default: ; ; Open Sound System Console Driver Configuration File ; [general] ; ; Automatically answer incoming calls on the console? Choose yes if ; for example you want to use this as an intercom. ; autoanswer=yes ; ; Default context (is overridden with @context syntax) ; context=default ; ; Default extension to call ; extension=s This is the important part of my extensions.conf: exten => 555,1,Dial(Console/dsp,20,A(trek)) exten => 555,2,Hangup John Bohman wrote: Tor, could you post the relevant config entries for your paging solution. Thanks in advance. JB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tor Roberts Sent: Thursday, August 26, 2004 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Overhead Paging I use a $20 soundcard on the asterisk server which is set to auto answer. Then connected to the soundcard is a an amplified horn. Whenever I dial extention 555, the soundcard picks up, plays the intercom tone from Star trek for a second to get everyone's attention, then then I start talkin over the horn. In your case, all you would need is a simple ampifier since you already have the speakers. Good luck, -Tor Roberts Deon Rodden wrote: I have several Cisco 7940's laying around, how do I piple the speakerphone through external speakers? I understand the amplifier part, but how do you get RCA/2.5mm outputs from the Cisco? For now, we just configured a "line 2" on all our phones with auto answer and, using the trick found in the wiki, when we page, every phone turns on and broadcasts whatever we say. However, there is an antiquated speaker system in the ceiling above that nobody knows anything about, it'd be neat to pipe a dedicated 7940 with Auto-Answer to the above Intercom. Rich Adamson wrote: I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a "Paging Unit" to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. If you search the archives I think you'll find this discussed several times. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
I use a $20 soundcard on the asterisk server which is set to auto answer. Then connected to the soundcard is a an amplified horn. Whenever I dial extention 555, the soundcard picks up, plays the intercom tone from Star trek for a second to get everyone's attention, then then I start talkin over the horn. In your case, all you would need is a simple ampifier since you already have the speakers. Good luck, -Tor Roberts Deon Rodden wrote: I have several Cisco 7940's laying around, how do I piple the speakerphone through external speakers? I understand the amplifier part, but how do you get RCA/2.5mm outputs from the Cisco? For now, we just configured a "line 2" on all our phones with auto answer and, using the trick found in the wiki, when we page, every phone turns on and broadcasts whatever we say. However, there is an antiquated speaker system in the ceiling above that nobody knows anything about, it'd be neat to pipe a dedicated 7940 with Auto-Answer to the above Intercom. Rich Adamson wrote: I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a "Paging Unit" to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. If you search the archives I think you'll find this discussed several times. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Echo
John, I have a dozen polycom ip 600 phones that are connected to the PSTN with two, digium 4 port fxo cards. I also had a lot of near side echo, but I do not believe that it came from the phones. I was able to get rid of it with these settings in my zapata.conf: echocancel=yes echocancelwhenbridged=yes echotraining=500 rxgain=8.0 txgain=0.0 The polycoms do have echo canceling hardware buit in, but by default only works on the speakerphone. You can enable it on the handset as well, but in my case, it made no difference because the echo came from the fxo cards. -Tor John Bittner wrote: Hi, Just install 6 new polycoms at a customer and all of them have a major echo issue. Have asterisk connected to the PSTN via digium 4 port fxo card in a P4 running fedora. I have tweaked zapata ran ztmonitor... just as a test I attached a cisco 7960 the cisco has no echo problems. Another issue I came up on is when I enable echo training I no longer can hear any inbound voice. Does anyone know if there is a setting in the polycom config that will cause this. The echo on the phones happen only on my side and only when I speak. The polycoms play back my voice but delayed. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI dead in USA?
Scott, I have an SBC BRI in California. I would not worry about them going away anytime soon. SBC just does not like to sell them because they want you to buy a PRI instead. The main thing to worry about is getting that BRI working with *. There are only a couple of cards that work the National ISDN, one of which is the Eicon Diva Sever cards. I tried hooking my BRI up to * a while ago, but was not successful. I was using a cheapo card, so I'm sure that was part of the problem. As far as callerid, I don't know if mine supports it. My BRI is connected to a portmaster, so I am not looking at callerid. Anyway, Good luck! Scott Stingel wrote: Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were "well, BRI is getting quite antiquated", and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 500 Voicemail
Wiley, I don't have any 500s, but I use 600s, which use the same file I think. Here is my digitmap: What this says is that if I dial 9, then a 7 digit local number, I don't need to hit send. If I dial 91, then 10 digit long distance number, I don't need to hit send. If I dial extension 85 plus any 2 digits ex., 8523, I don't need to hit send. If I dial extension 5,6, or 7 plus 2 digits, ex. 635, I don't need to hit send. And if I dial 9411, or 9911 (info or emergency) I don't need to hit send. Hope this helps. -Tor Wiley E. Siler wrote: I read the administrator document repeatedly. I have not been able to find a wiki that applied to digitmap feature at all and I have searched repeatedly and read several of the wikis regarding Polycoms. The administrators guide doesn't have enough context explanation to make the use of the digitmap understandable. That is the basis of my request for a digitmap explanation. I am not asking someone to write mine for me. I am asking to see an example and an explanation that gives context so I can write my own and know I have done it properly. My PBX is Asterisk and the setup is about as generic as generic can be. Polycoms over SIP to the PBX. If you know where the wiki is for digitmaps please send it. If you feel inspired, a short explanation of the relevance and context of digitmaps would be greatly appreciated. I know everyone has to take their own time to answer these emails and I truly appreciate that. That is why I do my research until I hit a wall, then I will ask here. I appreciate whatever you can spare time for. Thanks! Wiley -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any way to change ring back behavior for call park?
Hi, I have call center with 12 Polycom IP 600 SIP phones connected to *. When an incomming call comes in over the PSTN, 7 of those phones ring at the same time, and any of those 7 agents can pick up the new call. This works great. The problem is that if one of the agents parks a call and nobody picks up that parked call before the timeout (2 minutes), the parked call rings back all 7 extensions, just like a new incomming call. The agents want a way to identify a parked call ringing back. I have searched the archives and the wiki with no luck. What I would like to do would be to be able to set "ALERT_INFO" in park ring backs, so that I can force a different ring tone when the park times out. If that is not possible, I would like the parked call to only ring back to the extension that parked the call. If anyone has any good ideas, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 stops to send/receive calls
Jorge, That sounds strange to me. I have 12 IP 600s running without any of the problems that you are having. My first guess would be that they are configured wrong. Is the phone registering with asterisk? Is the phone dowloading it's config files from the FTP server? If you want to post your phone1.cfg and sip.conf, I can try to see if there is anything wrong with it. -Tor Jorge Mendoza wrote: Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls. After a reboot is works fine again. We have a * box with many BT101 and softphones working for months without any problem. I'm missing something? it is a bad config file? or it is a phone bug? Thank You for your time. Jorge Mendoza ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 600 Programmability
John, Oops! I was wrong, I do get a call waiting beep even if the call comes in on another extension. I still would prefer the phone to ring when another call comes in. If anybody knows how to do this, that would be great. -Tor Tor Roberts wrote: John, Mine gives the call waiting beep also, but only when a call comes in on the same extension that is in use. If a call comes in on another extension on my phone, then I get no beep, just a light on the button flashing and the sreen letting me know that there is another incoming call. -Tor -Tor John Baker wrote: My IP 600 gives me a "call-waiting" tone when another call comes in. I'm quite sure there's a setting for that in the xml. As for the feature buttons, I'll look at that this weekend, but it seems to me that using SIP with these phones precluded alot of the key programming stuff i.e., there was a chart in the admin guide that showed what the phone was capable of with respect to the keys under SIP and it wasn't much. John Tor Roberts wrote: Hi, Speaking of programming the IP 600, does anybody know how to programm any of the "feature" buttons to send a combination of digits while a call is in progress? The most obvious use would be to send "#700" while a call is in progress and label the button "Park". If I could do that, I would be very happy. Another option that would be nice would be if the phone would ring on a incoming when you are already on another line, instead of just flashing on the screen. Thanks, -Tor Roberts Erik Barker wrote: I would also be interested in similar functionality. We have agents using Polycom IP 600s that would like some sort of notification that they are logged into Asterisk Queues - either a flashing LED or perhaps some sort of graphic on the display. I know that there are numerous configuration options in the XML files and I've looked at the Admin guide, but i haven't seen any examples yet. Erik On Wed, 2004-06-16 at 04:44, Ray Burkholder wrote: I'm looking to program some sort of web-services function: user presses a button and send some info to a web server or scripting program. The web server or script returns some text and/or imagery for the screen. Lather, rinse, repeat. I saw in section 3.7.1 of the manual referenced below that there is a services function. However, it appears to not be enabled. Yet. Any other way of doing this, or has the 3.7.1 function been enabled yet? Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, June 15, 2004 11:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 600 Programmability Polycom IP 600's are fully programmable, much more so than the Cisco phones. Yes, you can program the phone buttons. That and just about everything else you can imagine is programmable via xml configuration files. Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00. pdf for the admin guide and you can see for yourself how great the difference is. John P.S. Here's the wiki: http://www.voip-info.org/wiki-Polycom+Phones Ray Burkholder wrote: Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing l
Re: [Asterisk-Users] Polycom IP 600 Programmability
John, Mine gives the call waiting beep also, but only when a call comes in on the same extension that is in use. If a call comes in on another extension on my phone, then I get no beep, just a light on the button flashing and the sreen letting me know that there is another incoming call. -Tor -Tor John Baker wrote: My IP 600 gives me a "call-waiting" tone when another call comes in. I'm quite sure there's a setting for that in the xml. As for the feature buttons, I'll look at that this weekend, but it seems to me that using SIP with these phones precluded alot of the key programming stuff i.e., there was a chart in the admin guide that showed what the phone was capable of with respect to the keys under SIP and it wasn't much. John Tor Roberts wrote: Hi, Speaking of programming the IP 600, does anybody know how to programm any of the "feature" buttons to send a combination of digits while a call is in progress? The most obvious use would be to send "#700" while a call is in progress and label the button "Park". If I could do that, I would be very happy. Another option that would be nice would be if the phone would ring on a incoming when you are already on another line, instead of just flashing on the screen. Thanks, -Tor Roberts Erik Barker wrote: I would also be interested in similar functionality. We have agents using Polycom IP 600s that would like some sort of notification that they are logged into Asterisk Queues - either a flashing LED or perhaps some sort of graphic on the display. I know that there are numerous configuration options in the XML files and I've looked at the Admin guide, but i haven't seen any examples yet. Erik On Wed, 2004-06-16 at 04:44, Ray Burkholder wrote: I'm looking to program some sort of web-services function: user presses a button and send some info to a web server or scripting program. The web server or script returns some text and/or imagery for the screen. Lather, rinse, repeat. I saw in section 3.7.1 of the manual referenced below that there is a services function. However, it appears to not be enabled. Yet. Any other way of doing this, or has the 3.7.1 function been enabled yet? Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, June 15, 2004 11:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 600 Programmability Polycom IP 600's are fully programmable, much more so than the Cisco phones. Yes, you can program the phone buttons. That and just about everything else you can imagine is programmable via xml configuration files. Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00. pdf for the admin guide and you can see for yourself how great the difference is. John P.S. Here's the wiki: http://www.voip-info.org/wiki-Polycom+Phones Ray Burkholder wrote: Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 600 Programmability
Hi, Speaking of programming the IP 600, does anybody know how to programm any of the "feature" buttons to send a combination of digits while a call is in progress? The most obvious use would be to send "#700" while a call is in progress and label the button "Park". If I could do that, I would be very happy. Another option that would be nice would be if the phone would ring on a incoming when you are already on another line, instead of just flashing on the screen. Thanks, -Tor Roberts Erik Barker wrote: I would also be interested in similar functionality. We have agents using Polycom IP 600s that would like some sort of notification that they are logged into Asterisk Queues - either a flashing LED or perhaps some sort of graphic on the display. I know that there are numerous configuration options in the XML files and I've looked at the Admin guide, but i haven't seen any examples yet. Erik On Wed, 2004-06-16 at 04:44, Ray Burkholder wrote: I'm looking to program some sort of web-services function: user presses a button and send some info to a web server or scripting program. The web server or script returns some text and/or imagery for the screen. Lather, rinse, repeat. I saw in section 3.7.1 of the manual referenced below that there is a services function. However, it appears to not be enabled. Yet. Any other way of doing this, or has the 3.7.1 function been enabled yet? Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, June 15, 2004 11:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 600 Programmability Polycom IP 600's are fully programmable, much more so than the Cisco phones. Yes, you can program the phone buttons. That and just about everything else you can imagine is programmable via xml configuration files. Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00. pdf for the admin guide and you can see for yourself how great the difference is. John P.S. Here's the wiki: http://www.voip-info.org/wiki-Polycom+Phones Ray Burkholder wrote: Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] not getting sound from chan_oss paging setup
Hi,
I am trying to setup an overhead paging system with asterisk. I have
followed some of the advice from the list and have oss.conf set for
autoanswer. The sound card and speakers work because they can play mp3s
just fine. When I call the extension, the asterisk console looks like
everything is working, but I get no sound. Here is what I get on the
console:
-- Executing Dial("SIP/509-4422", "CONSOLE/dsp") in new stack
<< Call placed to 'dsp' on console >>
<< Auto-answered >>
-- Called dsp
-- OSS/dsp answered SIP/509-4422
Is there something that I am missing? Any help would be appreciated.
-Tor
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Re: [Asterisk-Users] Polycom IP 600
Matt, I tried 1.2 out, but could not get multiple lines to register to the same sip channel. I am by no means an expert, so it is possible that somone else could figure out how to do it. Oh well, it would be nice if it worked. -Tor Tor Roberts wrote: Matt, Thank you very much, I will try the 1.2 release out as soon as I can. -Tor mattf wrote: Hello, the new Soundpoint 1.2 SIP release is on the Asterisk Polycom site for download: http://www.freedomphones.net/polycom/files/SoundPoint-IP_SIP_1.2.0.zip included with it is the new Admin Guide. Other older Polycom Soundpoint files are also available for download on the site: http://www.freedomphones.net/polycom/files/ Polycom Support will tell you that they don't send releases to non-certified partners, but the truth is that they are a hardware company and they will send you whatever releases you want if you either buy enough phones, go through a reseller that will support you(like ReviewVideo in Chicago) or get a contact far enough up the chain of command at Polycom to decide to give you the software. I was in talks with the VP of IP phone sales at Polycom for months and he said they were ready to work with the Asterisk community if a large reseller would step up to be the official reseller that they deal with in relation to the Asterisk community(they would even give us a contact in their engineering department). I was never able to get one of their resellers to agree to it without adding extra cost to the price of the phones. If someone is interested in taking up the cause and they have more free time than I do, send me an email and I'll let you know everything I know about it. Enjoy, MATT--- -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, June 14, 2004 10:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 600 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tor Roberts Sent: Monday, June 14, 2004 8:22 PM John, No, I have not tried 1.2, I did not know it was even out. Can this be downloaded from Polycom's site? If so, I will try it out. -Tor We had e-mailed Polycom about 2 months ago requesting that the ADA compliance (auto-reset Volume per call) be a feature as per the ADA specs rather than a requirement. Polycom wrote back and said it is in the 1.2 release. Polycom will only release the release to certified partners. John Baker wrote: Did you try the new sip firmware update? The latest is version 1.2 and has some fixes for what you're trying to do. John, do you care to share the new firmware? Unfortunately, like Tor, we do not have access to this either. What is the easiest way to get this release? When I call Insight (Where we purchased the phones) they have no clue what we are talking about. Thanks! - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 600
Matt, Thank you very much, I will try the 1.2 release out as soon as I can. -Tor mattf wrote: Hello, the new Soundpoint 1.2 SIP release is on the Asterisk Polycom site for download: http://www.freedomphones.net/polycom/files/SoundPoint-IP_SIP_1.2.0.zip included with it is the new Admin Guide. Other older Polycom Soundpoint files are also available for download on the site: http://www.freedomphones.net/polycom/files/ Polycom Support will tell you that they don't send releases to non-certified partners, but the truth is that they are a hardware company and they will send you whatever releases you want if you either buy enough phones, go through a reseller that will support you(like ReviewVideo in Chicago) or get a contact far enough up the chain of command at Polycom to decide to give you the software. I was in talks with the VP of IP phone sales at Polycom for months and he said they were ready to work with the Asterisk community if a large reseller would step up to be the official reseller that they deal with in relation to the Asterisk community(they would even give us a contact in their engineering department). I was never able to get one of their resellers to agree to it without adding extra cost to the price of the phones. If someone is interested in taking up the cause and they have more free time than I do, send me an email and I'll let you know everything I know about it. Enjoy, MATT--- -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, June 14, 2004 10:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 600 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tor Roberts Sent: Monday, June 14, 2004 8:22 PM John, No, I have not tried 1.2, I did not know it was even out. Can this be downloaded from Polycom's site? If so, I will try it out. -Tor We had e-mailed Polycom about 2 months ago requesting that the ADA compliance (auto-reset Volume per call) be a feature as per the ADA specs rather than a requirement. Polycom wrote back and said it is in the 1.2 release. Polycom will only release the release to certified partners. John Baker wrote: Did you try the new sip firmware update? The latest is version 1.2 and has some fixes for what you're trying to do. John, do you care to share the new firmware? Unfortunately, like Tor, we do not have access to this either. What is the easiest way to get this release? When I call Insight (Where we purchased the phones) they have no clue what we are talking about. Thanks! - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 600
John, No, I have not tried 1.2, I did not know it was even out. Can this be downloaded from Polycom's site? If so, I will try it out. -Tor John Baker wrote: Did you try the new sip firmware update? The latest is version 1.2 and has some fixes for what you're trying to do. John On Mon, 2004-06-14 at 14:21, Tor Roberts wrote: Eric, I tried this and could not get it to work. What I ended up doing is giving each button a different extension and then set the phone to "divert" to the second extension on busy. This is done with the "busy divert.busy.1.enabled=" and "divert.busy.1.contact=" tags in the phones config file. This has some drawbacks in that you have to make way more extensions, but you also have to use call waiting on each line. If there is a better way to do it, I would like to hear it. But this does get the job done. It would be great if you could just register all the lines to the same extension. -Tor Roberts Eric Mandel wrote: I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if anyone can confirm that this works with the polycoms. I know the 7960s support this, but I want to make sure the Polycom sales team wasn't just saying Yes to make the sale. Any comments are appreciated. -Eric -Original Message- Subject: fwd on busy when calling multiple extensions at once Chris A. Icide wrote: IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 600
Eric, I tried this and could not get it to work. What I ended up doing is giving each button a different extension and then set the phone to "divert" to the second extension on busy. This is done with the "busy divert.busy.1.enabled=" and "divert.busy.1.contact=" tags in the phones config file. This has some drawbacks in that you have to make way more extensions, but you also have to use call waiting on each line. If there is a better way to do it, I would like to hear it. But this does get the job done. It would be great if you could just register all the lines to the same extension. -Tor Roberts Eric Mandel wrote: I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if anyone can confirm that this works with the polycoms. I know the 7960s support this, but I want to make sure the Polycom sales team wasn't just saying Yes to make the sale. Any comments are appreciated. -Eric -Original Message- Subject: fwd on busy when calling multiple extensions at once Chris A. Icide wrote: IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
Chris, As far as the Cisco phones, they are not an option as I already have the Polycoms. The Ciscos are overpriced anyway. It was my understanding that asterisk would not let you register the same extension more than once. If that is not the case, I will try to register the same extension to all 6 lines. If all 6 lines are used on any of the phones then I imagine that only the other 3 phones will ring. I don't think this will happen, as I only have 8 incoming lines. If it does become a problem, then I could enable call waiting on the last 2 lines so that each phone can handle 8 calls. Thank you for your advice! -Tor Roberts Chris A. Icide wrote: IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: > >You might consider using the Cisco SIP phones. They're smart enough to >accept incoming calls for as many call appearances you have with the >same SIP registration. > >-brian > >Tor Roberts wrote: > >> Hi, >> I am setting up a dispatch center where will have 4 call takers, all >> with Polycom IP 600 Sip phones. Each phone will be setup with 6 >> extensions each. When a new call comes in, the first extension on all >> the phones will ring. This works fine, the problem is when one of the >> dispatchers is already using her first extension and another call >> comes in. What happens now is that the remaining 3 phones ring on the >> first extension, but the dispatcher who is on a call, her phone does >> not ring. I want her second extension ring along with the other 3 >> phones first extensions. >> >> In sip.conf I have all the extensions set to incominglimit=1 and the >> pertinent part of extensions.conf is: >> >> exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) >> exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) >> >> and so on. >> >> If anybody has any insight, or a better solution, that would be great. >> >> Thanks, >> >> -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fwd on busy when calling multiple extensions at once
Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN -> card? -> Asterisk
Michael, I won't be much help because I am just a couple of steps ahead of you, but I will try. It looks like there are not many people in the U.S. using BRI. Because it is so unpopular here, there are not many cards available that work. The key to getting a card that "might" work here, is if the card supports the NI-1 protocol. Most cards that do support it are active cards like the Eicon Diva Server, and are not cheap. There are a few cheap passive cards that support NI-1, but I don't know if they work. Another problem with most of the cards that do work, is that they don't have a U interface, so you need to buy an external NT1, which will give you the required U interface. There is a company in Austalia that makes a card that does NI-1 and has a U interface, but I don't know if anyone has used it. The url for their card is: http://www.traverse.com.au/productview.do?product_id=14 I just picked up a cheap Dynalink card on ebay which I am going to try on Monday, when I can plug it into a BRI line. I am not holding my breath though. My suggestion is to get a more expensive, active BRI card, and see if it works. If you have any luck, please let me know. -Tor Michael Welter wrote: I'm having trouble determining which ISDN4Linux devices are usable in the US. I want to integrate ISDN into my Asterisk PBX. My circuit provider is Qwest. Does anyone have a working ISDN BRI interface in the US? Does the fax work? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any ISDN BRI card recommendations for North America?
Rob, Thanks for the info. Since it seems like BRI is not too popular in the U.S., I think that I will try to pick up a DIVA PCI and see if it will work with CAPI or i4l. -Tor Roberts Rob Fugina wrote: On Wed, Mar 17, 2004 at 12:25:49PM -0800, Tor Roberts wrote: Hi all, I have been using Asterisk for a couple of months now with some GS handsets and an X100P FXO card. The system works great, but I would like to add ISDN BRI to take advantage of the extra features, faster call setup time, etc. I was wondering if anyone could recommend any BRI cards that work in the U.S. and don't cost a fortune. I have checked the archieves and it seems like there are not many people in the U.S. using BRI. I was hoping that I could use either an Eicon DIVA PCI or an Eicon DIVA PRO as they are not that too expensive. If anyone has used either of these cards in the U.S., or can recommend another alternative, that would be great! I'm looking for the same thing, for the same reasons. I believe the PRO version is not supported by the i4l drivers. Someone referred me to the following product, made by an Australian company it seems, but I haven't been able to find a source... To be honext, I haven't tried too hard -- busy with other things. http://www.traverse.com.au/productview.do?product_id=14 Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any ISDN BRI card recommendations for North America?
Hi all, I have been using Asterisk for a couple of months now with some GS handsets and an X100P FXO card. The system works great, but I would like to add ISDN BRI to take advantage of the extra features, faster call setup time, etc. I was wondering if anyone could recommend any BRI cards that work in the U.S. and don't cost a fortune. I have checked the archieves and it seems like there are not many people in the U.S. using BRI. I was hoping that I could use either an Eicon DIVA PCI or an Eicon DIVA PRO as they are not that too expensive. If anyone has used either of these cards in the U.S., or can recommend another alternative, that would be great! Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
