Re: [Asterisk-Users] SIP Asterisk Polycom Reinvite
I had a one way audio problem with my Polycom 501's and it turned out that the cord wasn't plugged in to the handset all the way. It looked like it was in, but it wasn't in all the way till it clicked. Matt Damon Estep wrote: Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are canreinvite=yes and the PSTN termination path is canreinvite=no then calls between polycoms should not have asterisk in the media stream and wan link utilization is reduced. The problem looks like the Polycom keeps trying to reinvite the sonus and the call never sets up right, and not with all calls… Any experience with this? Maybe there is a totally different issue I am overlooking? About 3 to 5% of all Polycom to PSTN via asterisk>SIP peer calls are impacted. I have not set the Polycom canreinvite=no yet, hoping to not have to do that as the wan link is a t1 that is also used for data. Thanks for any help! Damon ___ --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] metermaid patch
Dr. Michael J. Chudobiak wrote: I'd like to be able to use my Snom 360 LEDs to view the status of parking slots, so I'm trying to install the "metermaid" patch (http://bugs.digium.com/view.php?id=5779). Can someone help an svn newbie figure out how to install this patch? I've done the following: Any update on this? Also, is there any chance that the metermaid functionality will be added to Asterisk? Matt ___ --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] Good VoIP providers that support Asterisk PBX's
Dovid Bender wrote: I personaly use VoipJet, Teliax and myPhoneCompany. They are all great. Dont remember if teliax supported IAX. I know that myPhoneCompany for sure dosent. They use SIP. I did however ind that thier voice quality is very good. I'm sorry, but you don't remember if Teliax supports IAX? They most certainly do, look at the name... Tel-IAX ;-) Matt ___ --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] FW: auto provision of IP501 polycom
I have the same problem. I'm running CentOS, which comes with vsftpd, do you know of anyway to do this using vsftpd? Thanks, Matt Noah Miller wrote: Hi Again Damon - I just remembered that the FTP server setup can be tricky, too. The default username has capitalized letters, and this doesn't work with a lot of FTP servers. I had to use ProFTPd to get it done. I created a user account called plcmspip, and added the following to /etc/proftpd.conf (or wherever you choose to put your config file): UserAlias PlcmSpIp plcmspip - Noah -- Forwarded Message From: Noah Miller <[EMAIL PROTECTED]> Date: Thu, 23 Feb 2006 11:34:31 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: <[EMAIL PROTECTED]> Conversation: auto provision of IP501 polycom Subject: Re: auto provision of IP501 polycom Hi Damon - Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? Sure, works great! I'm not sure if you got the TFTP config from the gentleman who suggested it, but this is really dependent on what DHCP server you are using. For example, we use Cisco routers, and the option to add is: option 66 ascii "xxx.xxx.xxx.xxx" where the xxx's are the IP address (I don't think DNS names will work). ISC's DHCP server is a little different: option tftp-server-name "XXX.XXX.XXX.XXX"; Yes, it should be tftp-server-name, even if you use FTP (or HTTPS, I believe). - Noah -- End of Forwarded Message ___ --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: Remapping Polycom IP501 buttons
Henry Kwan wrote: Hi Noah, You've run into the same problem a lot of other people have had. Remapping hard keys works fine, but remapping soft keys does not. In fact, trying to remap the soft keys results in some pretty weird behavior. The Polycom manual is a little misleading in that it doesn't mention this at all. My best guess is that the softkeys don't work because they can mean different things depending on what the phone is doing at the time. Polycom, if you're reading this, this would be another great feature to have! Thanks for the info. That would explain a lot. The manual clearly states that SpeedDial should work though. On page 114 of the admin guide, it says that "key.x.y.subPoint.prim" will "Sets the sub-identifier for key functions with a secondary array identifier such as SpeedDial." But when I try to set it: I get a volume-up action instead. So I guess it's a bug that they haven't gotten to fixing yet? I have had the same exact problem a button remapping gone wrong that results in volume up. I don't know if it's a bug, or bad documentation or what, but it's very frustrating. Matt ___ --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: Remapping Polycom IP501 buttons
Noah Miller wrote: Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: This works as expected but if I try to change the remapping to any other value like "MyStatus", "SpeedDialMenu", or "BuddyStatus", it doesn't work. I got the list of values from Polycom's admin guide. Why does "DoNotDisturb" work and no other values that I've tried? You've run into the same problem a lot of other people have had. Remapping hard keys works fine, but remapping soft keys does not. In fact, trying to remap the soft keys results in some pretty weird behavior. The Polycom manual is a little misleading in that it doesn't mention this at all. My best guess is that the softkeys don't work because they can mean different things depending on what the phone is doing at the time. Polycom, if you're reading this, this would be another great feature to have! Who has hard buttons remapped for anything but the simplest of actions? If you do, I would very much like to hear about it. Can you post some details? Matt ___ --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] Remapping Polycom IP501 buttons
Henry Kwan wrote: Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: This works as expected but if I try to change the remapping to any other value like "MyStatus", "SpeedDialMenu", or "BuddyStatus", it doesn't work. I got the list of values from Polycom's admin guide. Why does "DoNotDisturb" work and no other values that I've tried? This is the big question as far as I'm concerned with using Polycomm phones, I have about 30 501's running in our office and I like everything about them except the button remapping problem. If someone can figure that I would be totally psyched. Like you, I have been able to do simple remaps, like setting the Transfer button to issue a #, but anything more complex than that just doesn't work. Matt ___ --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] Polycom IP301: Pass-through ethernet port unusable?
What version of the firmware? Jerry Glomph Black wrote: Have just done a deployment of 45 of these puppies. They are doing their main job quite well, but of course there are minor kinks. A not-so-minor one is that if one attempts to plug a PC into the 2nd RJ-45 jack, as soon as you send any reasonable amount of traffic (even casual web surfing) the phone seizes. We had to run a bunch of cables in a big rush to users' PCs, having (erroneously) believed that the passthru RJ45 would be a usable port! Has anyone out there experienced this? ___ --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] Polycom 501 remapping keys
What I really want to be able todo is use the services button or any of the other buttons that serve no purpose right now. I would like to have it start a page (which on my * box is just dialing a particular extension), I have this working on my Polycom 501's using the 3rd line appearance, however I would rather keep that a line appearance. I would also like to be able to use a button to park a call. My users can usually get it right at this point, but they still mess it up far too often. Matt Bill Gibbs wrote: Yeah I just got in a 301 to test and I can configure a key (for example in sip.cfg key.IP_300.2.function.prim="Messages"/> and then when I hit the line 2 key it drops me right into VM (since I have that configured too) Still playing around, I noticed that if you get the soft keys (the menu ones under the LCD) then it ALWAYS is that function... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Sent: Friday, December 09, 2005 9:06 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 remapping keys There has been a fair amount of converstaion about this, but I'm not sure anyone really has this working. I had exactly the same problem that the button got remapped to a volume up function. The only button remapping I got working was to map the Transfer button to the # key so that when you hit Transfer it started and Asterisk based transfer. I would love to hear from someone who has this working. Matthew O'Connor [EMAIL PROTECTED] wrote: I've tried to configure the "services"-key on my Polycom 501 to run a SpeedDial-entry in [MACADRESS]-directory.xml (which would call a asterisk-extension that starts SayUnixTime) but i have not been able to accomplish my goal. Whe configuring the SpeedDial-function in sip.cfg "VolUp" is started when i press the Services-Key. Also some other possible functions listed under 4.6.1.15 in the SIP 1.6 Administrator Guide fail. Some of them were working with the expected function, some where not giving any response at all but some where starting totally different functions, e.g. configuring "Redial" as the function starts "Settings", function "Messages" starts "Redial", "SpeedDialMenu" starts "VolUp", "VolUp" starts "Line1" :-[ I've seen that other failed as well (http://lists.digium.com/pipermail/asterisk-users/2005-October/130129.ht ml) - anyone ever got this working? Maybe with BootROM 3.0/3.1? Or should i downgrade to 1.5 where there was a ipmid-file for remapping-info...? I'm running Firmware 1.6.2.0041/BootROM 2.6.2.0032 regards Christian ___ --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] 1.6.3 Polycom Firmware?
Kevin Ragsdale wrote: Has anyone tried the newest Polycom firmware? The release notes indicate they have added support for a new BLA draft. New BLA draft? Would you mind explaining what that is. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Softkeys & Voicemail Button
Sean Cook wrote: I would like to alter some of the softkey options on the phone, such as removing Buddies/MyStat buttons, and possibly replacing them with other options -- as well as altering the transfer/conference softkey options for situations where a call is in progress. I have read through the Admin Guide, but still remain unsure on how to actually make these changes -- and what configuration options to use. Has anyone does this before, if so, do you have the configuration sections that need to be modified? As far as I can tell, nobody has figured this out. There are however many people on this list who would be VERY interested in getting these questions answered. Not so... They do require ftp or tftp configuration: To remove the Buddies/MyStat soft buttons set both enabled flags here to 0 Right, I figured out how to get my messages button working so that when you press it you get sent directly into the VM system, however I have not been able to figure out how to remap any of the soft keys, or the other buttons such as the services button. There is a little documentation in the admin guide which allowed me to reprogram the transfer button to be the equivilent of at # keypress, but there is more I would like todo. Remapping the other buttons is what nobody has figured out yet. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Softkeys & Voicemail Button
Brian wrote: I would like to alter some of the softkey options on the phone, such as removing Buddies/MyStat buttons, and possibly replacing them with other options -- as well as altering the transfer/conference softkey options for situations where a call is in progress. I have read through the Admin Guide, but still remain unsure on how to actually make these changes -- and what configuration options to use. Has anyone does this before, if so, do you have the configuration sections that need to be modified? As far as I can tell, nobody has figured this out. There are however many people on this list who would be VERY interested in getting these questions answered. Also, I have noticed that when multiple registrations are placed onto a Polycom Phone, the phone then presents you with the "Message Center" when you press the Messages button. Is it possible to configure the phone to always place you in the VMB of the first line registration, as the phone does if you only have a single registration in progress? I don't know, I only use one single registration. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 files
Jerry Jones wrote: Has anyone been able to get Polycom phones to use a different directory when it creates mac-phone.cfg and mac-directory.xml instead of placing them in the root directory. similar to specifying a log directory which does work. Not to my knowledge, but that would be a nice feature. It's my impression that it only creates these files if it can't find config files that you create for it, is that right? Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 501 can't find TFTP server?
Doug wrote: Got a couple of brand new Polycom 501s in. As usual, set them up to TFTP to get new firmware. Couldn't find the server. Phones can sense Ethernet presence fine. I even set up a FTP account on a different server and put the firmware files there. Couldn't find that server either. So you have other Polycom phones that do work? Also, are you sure they are getting all the DHCP server information they need? Sounds to me like you aren't getting the boot-server setting from the DHCP server. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: Polycom IP501 and record on demand
Mojo with Horan & Company, LLC wrote: That is perfect for one-button remaps! I guess I migrated away from one-button features in * but I see the light now. Yeah, I have this working for one button remaps (remapping to the transfer button to #) but I never got the SpeedDial trick working. Could you resend me your exact I do it it remaps to volume up which is really annoying. The trouble Matthew and I were having was to stimulate presses of more than one button in a sequence -- "SpeedDial" function was the only one I could find that was close, but this opens a new call appearance for the call rather than just playing the dtmf over the open one. This would be wonderful if anyone could get this working. The really frustrating thing is that Polycom lists the "reprogramable buttons" as one of their selling points. Anyway, I have given up for now. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 IP501 and record on demand
Jonathan k. Creasy wrote: I probably can't provide any better information for you, however, have you looked through the Polycom configuration files. The button mappings are there. I haven't spent much time with it so I can not attest to what you can map them to do. I have spent a lot of time looking through the config files, and also through the Admin guide which does show you how to remap a button (and I have it working for simple one button remaps), and it give you a list of functions that you can use while remapping buttons, but it gives you no infomratoin about those functions, what they do, or specifically how they work. This is the really annoying part, is that they give you enough info to get really close, but not enougth to actually make it work. Oh well. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 IP501 and record on demand
Matt Gibson wrote: You could also take a look at features.conf, and use ** for blind transfers, ## for attended transfers, *0 for recording, and *1 to hangup. I haven't tried mapping them to polycom buttons, but there was recently a discussion about that, just this week you can search the archives. There was a discussion (of which I was a part of) however there was no resolution. I have not found any good documentation on how to remap Polycom buttons. At this point I'm willing to pay for some help. Anybody got some better info on this? Thanks, Matthew O'Connor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Middle Ground between POTS and T1?
I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but smaller. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Voicemail -> new feature request
Kib Eki wrote: It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Also, one simple thing. Is it possible to listen to my greetings without having to re-record them? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Button Remapping: Part 2
Thanks to some help from this list, I can remap useless buttons (such as services). However I can only remap to other things that are also useless. So my question is: Does anyone have more detail on how to use the functions listed in the Admin Guide in section 4.6.1.15. I would like to be able to dial a number from these buttons, or perhaps perform a series of key presses, but so far I haven't got that working. Is there some document somewhere that details these functions, or at least provides some examples? Thanks, Matthew O'Connor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Button Remapping, HELP!
While I don't have it working yet, I think I have it figured out. I have to add entries to my sip.conf Based on your example I was able to find the relevant info in the Polycom SIP 1.5 Admin Guide section 4.6.1.15. My next question, which I haven't found in the admin guide (at least not yet) is where to you get a list of the buttons and their respective numbers? Thanks again, Matthew Mojo with Horan & Company, LLC wrote: Do you already have an block in your sip.cfg? add the in there: Try putting: ... ... key.IP_500.31.subPoint.prim="3"/> Moj Matthew T. O'Connor wrote: Ok, that would be helpful for me with some other problems, however I don't see "I'm using the 1.5.2 Sip firmware the the conf files that came with that, so I don't have an ipmid.cfg file. Is this something I can just add to my sip.conf? Anyone out there any suggestions on how to do the speed dial "in-call"? Thanks, Matt Mojo with Horan & Company, LLC wrote: Matthew, when I tried this, I couldn't get the soundpoints to dial in-call. They thought there were picking up a new line for a new call. I created a speed-dial entry (in MACADDRESS-directory.xml, "Park#70#3") and then in ipmid.cfg: key.IP_500.31.subPoint.prim="3"/> This tells the phone to run Speed Dial 3 whenever the Services button (button #31 on a 500/501) is pressed. I hope someone can help us configure them now to dial these digits in-call... Mojo Matthew T. O'Connor wrote: I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial #70#. Or if I could add a soft button during a call that would work too. Anyone have any suggestions on how to do this? Thanks much, Matthew O'Connor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Button Remapping, HELP!
Ok, that would be helpful for me with some other problems, however I don't see "using the 1.5.2 Sip firmware the the conf files that came with that, so I don't have an ipmid.cfg file. Is this something I can just add to my sip.conf? Anyone out there any suggestions on how to do the speed dial "in-call"? Thanks, Matt Mojo with Horan & Company, LLC wrote: Matthew, when I tried this, I couldn't get the soundpoints to dial in-call. They thought there were picking up a new line for a new call. I created a speed-dial entry (in MACADDRESS-directory.xml, "Park#70#3") and then in ipmid.cfg: key.IP_500.31.subPoint.prim="3"/> This tells the phone to run Speed Dial 3 whenever the Services button (button #31 on a 500/501) is pressed. I hope someone can help us configure them now to dial these digits in-call... Mojo Matthew T. O'Connor wrote: I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial #70#. Or if I could add a soft button during a call that would work too. Anyone have any suggestions on how to do this? Thanks much, Matthew O'Connor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom: Button Remapping, HELP!
I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial #70#. Or if I could add a soft button during a call that would work too. Anyone have any suggestions on how to do this? Thanks much, Matthew O'Connor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ParkAndAnnounce Question
I have an office up and running with 40 SIP handsets. Currently when an incoming call is parked, they then dial ext 9876 which I have setup to do a page (it does this through an agi script that uses the Polycom autoanswer to page via the phones). What I would like to be able to do is transfer the call to a new extension #699 for example, and have that park the call and then connect the parker to the paging extension so that they can announce to the office that someone is parked. Any thoughts on this? Thanks, I am trying to figure out A) if this is possible, and assuming it is, B) How do I do it. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 501: takes calls, but fast busy on dial out?
Are you sure there is no firewall on the server? Or are you sure that the Polycom phone is trying to connect to the * server on the expected port number? What happens when you try to telnet to the SIP port on the * box? Matt Doug wrote: Hi, Has anyone seen this before? The phones are registered OK, and they can take incoming calls, but all I get is a fast busy whatever I dial. I've tried regular numbers, *98, etc. Looking at the Asterisk Command Line Interface, I don't see any text outputted when I try to dial out. I wonder if it's even getting to the Asterisk server. Where does it get the fast busy from--inside the phone? Another clue is the Flash Operator Panel. The extensions show up, but the "devices" are greyed out. Here is what "sip show peers" has: Name/usernameHostDyn Nat ACL Mask Port Status 15102/15102 XXX.XXX.XXX.XXX D N 255.255.255.255 1044 OK (69 ms) 15101/15101 XXX.XXX.XXX.XXX D N 255.255.255.255 1043 OK (69 ms) Do the ports matter? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this normal?
Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line. Nearly every hour, almost on the hour I get this: Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/3 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/4 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/5 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/6 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/7 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/8 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/9 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/10 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/11 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/12 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/13 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/14 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/15 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/16 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/17 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/18 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/19 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/20 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/21 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/22 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/23 successfully restarted on span 1 Is this normal? Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Setup Questions
Matt Gibson wrote: Matthew T. O'Connor wrote: I'm not sure I know about 3, where can I read more about the Polycom's "known issues". Are you talking about the problems with type=friend? Thanks for you help. Matt Hey Matt, This is what I was referring to from the wiki: http://voip-info.org/tiki-index.php?page=Polycom%20SoundPoint%20IP%20500 look for "phone randomly freezes up" although, I'm not sure that's your issue anymore. Nope not my issue, just to solve the mystery for everyone. I did fix the problem. Jonathan nailed it on the head when he mentioned the loose headset cable. A few of my phone didn't have the headset cable plugged all the way (till they clicked), but were in enough to work until they slid out just a little. The two things that threw me off track were 1) the person could hear but not talk, I would tend to think that if you headset works one way, it should work the other way too, apparently not; and 2) The headset diagnositc passed, so I guess the cable was back "in" while I did the test. Anyway, I failed to pass the "is it plugged in" question :-) Thanks for you help. Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Setup Questions
Jonathan k. Creasy wrote: I had a loose headset cable doing that one day Can't be that, cause the hardware diagnostics worked fine. So it doesn't appear to be defective hardware, or loose cables or anything like that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Setup Questions
Doug wrote: At 14:40 9/27/2005, Matthew T. O'Connor, wrote: >So while I'm waiting to see if anyone can help with those questions, I >thought I would ask one more :-) > >All of the sudden 3 of my Polycom501 handsets started having a 1 way >audio problem. Did you make certain "canreinvite" equals "no"? Here is the entry in sip.conf for one of the problem phones. [124] username=124 type=friend secret=124 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=24 East Sales Office <124> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Setup Questions
Matt Gibson wrote: Matthew T. O'Connor wrote: The 3 phones in question were working morning yesterday, then for no apparent reason, the user could no longer talk. The polycom user could hear the person at the other end, but could not talk to them. Nothing has changed as far as I can tell, and I have no idea even where to start looking. Could be one of three things: 1 - Codec Problems 2 - NAT (but you mention no firewall and same LAN, so not this one) 3 - Polycom's known problems I'm willing to bet 3. Try rebooting the phone and seeing if it works for you then. I have rebooted all three phones a few times, it makes no difference. I have rebooted the phones both from the phone itself (+,-,Hold,Messages) and from the sip notify command in Asterisk itself. I'm not sure I know about 3, where can I read more about the Polycom's "known issues". Are you talking about the problems with type=friend? Thanks for you help. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Setup Questions
So while I'm waiting to see if anyone can help with those questions, I thought I would ask one more :-) All of the sudden 3 of my Polycom501 handsets started having a 1 way audio problem. My setup: 30 Polycom501 handsets all connected to Asterisk (CVS HEAD from a week or two ago) over a 100Mb LAN. The * server isn't using a firewall. What Happened: The 3 phones in question were working morning yesterday, then for no apparent reason, the user could no longer talk. The polycom user could hear the person at the other end, but could not talk to them. Nothing has changed as far as I can tell, and I have no idea even where to start looking. Also I did the on phone Diagnositics and the handset is working according to that. Any help would be greatly appreciated. Thanks, Matthew O'Connor Matthew T. O'Connor wrote: OK I have just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't think it works with Asterisk. Is this true? Is there a way to integrate the to call parking system etc? 1a) If I can't use the Polycom built-in call park feature, is there a way to remap one of the buttons on the left (the services button for example) to dial *70 for my users? 2) Transferring Calls: They way our office operates, I would prefer the default transfer method to be a blind transfer. Is there a way to reprogram the Polycoms to default to blind transfers? There are more questions but that is all for now :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Setup Questions
OK I have just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't think it works with Asterisk. Is this true? Is there a way to integrate the to call parking system etc? 1a) If I can't use the Polycom built-in call park feature, is there a way to remap one of the buttons on the left (the services button for example) to dial *70 for my users? 2) Transferring Calls: They way our office operates, I would prefer the default transfer method to be a blind transfer. Is there a way to reprogram the Polycoms to default to blind transfers? There are more questions but that is all for now :-) Thanks, Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Reboot Script
Kristian Kielhofner wrote: Matthew T. O'Connor wrote: Isn't sip_notify.conf just an Asterisk 1.2 thing? I'm running 1.0.9. I'm trying to setup a production system for my company, do you think 1.2 is ready for that? It sure is! You should be testing it! :) Test it and see, but 1.2 will be "STABLE" pretty soon here... I'm happy to help out and test out 1.2 beta, but I don't think "pretty soon" will be soon enough. We are opening our new office in less than two weeks. I can't imagine that 1.2 will be out of Beta by then. Thanks for you help. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Reboot Script
Kristian Kielhofner wrote: Matthew T. O'Connor wrote: Any Ideas? Have a look at /etc/asterisk/sip_notify.conf look for: [polycom-check-cfg] So, from the CLI: asterisk -r sip notify polycom-check-cfg [name] Isn't sip_notify.conf just an Asterisk 1.2 thing? I'm running 1.0.9. I'm trying to setup a production system for my company, do you think 1.2 is ready for that? Thanks, Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Reboot Script
Hello, I'm trying to setup the revised Polycom remote reboot script as
found on:
http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script
I'm not sure how to use this script, it's just a perl script, so I tried
creating an executable perl script and running it, but I get the following:
[EMAIL PROTECTED] agi-bin]# ./polycom_reboot.pl 192.168.3.205
Checking ARP table.
192.168.3.205 is reachable.
checking for polycom config name...
touching config file /home/polycom/0004f201d398.cfg
Use of uninitialized value in concatenation (.) or string at
./polycom_reboot.pl line 97, line 3.
Use of uninitialized value in concatenation (.) or string at
./polycom_reboot.pl line 99, line 3.
Use of uninitialized value in concatenation (.) or string at
./polycom_reboot.pl line 99, line 3.
reboot of phone 192.168.3.205 was successful
While it does say it is successful, I can tell you the phone does NOT
reboot.
line 97 looks like this:
$call_id = $tm . "msgto$sip_to";
It's part of this sub routine:
sub reboot_sip_phone {# Send the phone a check-sync to reboot it
$phone_ip = shift;
$local_ip = shift;
$sip_to = shift;
$sip_from = "asterisk";
$tm = time();
$call_id = $tm . "msgto$sip_to";
$httptime = `date -R`;
$MESG = "NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP $local_ip
From:
To:
Event: check-sync
Date: $httptime
Call-ID: [EMAIL PROTECTED]
CSeq: 1300 NOTIFY
Contact:
Content-Length: 0
";
Any Ideas?
Thanks,
Matt O'Connor
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[Asterisk-Users] Audio Problem when zaptel modules are loaded
Hello, I have an Asterisk @ Home box up and running. I'm trying to
prepare the box to be the phone system in our new office.
I have 3 Polycom 500's with the latest SIP firmware and all is fine, I
can call Asterisk, check my voicemail, call the other handsets, transfer
call, even make calls to the PSTN via voipjet. So I'm doing ok. The
problem I have now is that when I reboot into a kernel that has the
zaptel modules compiled for it, then I can no longer call asterisk
server and hear anything. I can however still call the PSTN via voipjet
and I can call the other phones.
When you look at the * log output below the only thing that I see as
potentially relevant is the fact that when I can't hear the vm-login,
there is a "Scheduling timer at 160 sample intervals" just before it,
but when it works, there is not.
Any ideas? Thanks,
Matt
1.
This first log excerpt is when I boot into a linux kernel that has
the zaptel drivers installed, this results in asterisk not being
able to play sound to me, so I can't hear the vm-login recording.
2.
Aug 26 15:58:18 DEBUG[2856]: Setting NAT on RTP to 0
3.
Aug 26 15:58:18 DEBUG[2856]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Found
4.
Aug 26 15:58:18 DEBUG[2856]: Setting NAT on RTP to 0
5.
Aug 26 15:58:18 DEBUG[2856]: Check for res for 201
6.
Aug 26 15:58:18 DEBUG[2856]: Call from user '201' is 1 out of 0
7.
Aug 26 15:58:18 DEBUG[2856]: build_route: Contact hop:
8.
Aug 26 15:58:18 VERBOSE[2856]: -- Executing
Answer("SIP/201-ec07", "") in new stack
9.
Aug 26 15:58:18 VERBOSE[2856]: -- Executing
Wait("SIP/201-ec07", "1") in new stack
10.
Aug 26 15:58:18 DEBUG[2856]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 2: Found
11.
Aug 26 15:58:19 VERBOSE[2856]: -- Executing
VoiceMailMain("SIP/201-ec07", "default") in new stack
12.
Aug 26 15:58:19 DEBUG[2856]: Ooh, format changed from unknown to ulaw
13.
Aug 26 15:58:19 DEBUG[2856]: Scheduling timer at 160 sample intervals
14.
Aug 26 15:58:19 VERBOSE[2856]: -- Playing 'vm-login' (language
'en')
15.
16.
This next excerpt is from a different kernel on the same box that
doesn't have the zaptel driver compiled, here the sound works
find, and I can here the vm-login recording:
17.
Aug 26 16:07:05 DEBUG[2820]: Manager received command 'Command'
18.
Aug 26 16:07:05 DEBUG[2820]: Manager received command 'Command'
19.
Aug 26 16:07:31 DEBUG[2820]: Setting NAT on RTP to 0
20.
Aug 26 16:07:31 DEBUG[2820]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Found
21.
Aug 26 16:07:31 DEBUG[2820]: Setting NAT on RTP to 0
22.
Aug 26 16:07:31 DEBUG[2820]: Check for res for 205
23.
Aug 26 16:07:31 DEBUG[2820]: Call from user '205' is 1 out of 0
24.
Aug 26 16:07:31 DEBUG[2820]: build_route: Contact hop:
25.
Aug 26 16:07:31 VERBOSE[2820]: -- Executing
Answer("SIP/205-6859", "") in new stack
26.
Aug 26 16:07:31 VERBOSE[2820]: -- Executing
Wait("SIP/205-6859", "1") in new stack
27.
Aug 26 16:07:31 DEBUG[2820]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 2: Found
28.
Aug 26 16:07:32 VERBOSE[2820]: -- Executing
VoiceMailMain("SIP/205-6859", "default") in new stack
29.
Aug 26 16:07:32 DEBUG[2820]: Ooh, format changed from unknown to ulaw
30.
Aug 26 16:07:32 VERBOSE[2820]: -- Playing 'vm-login' (language
'en')
31.
Aug 26 16:07:35 WARNING[2820]: Couldn't read username
32.
Aug 26 16:07:35 VERBOSE[2820]: == Spawn extension
(from-internal, *98, 3) exited non-zero on 'SIP/205-6859'
33.
Aug 26 16:07:35 VERBOSE[2820]: -- Executing
Macro("SIP/205-6859", "hangupcall") in new stack
34.
Aug 26 16:07:35 VERBOSE[2820]: -- Executing
ResetCDR("SIP/205-6859", "w") in new stack
35.
Aug 26 16:07:35 DEBUG[2820]: cdr_mysql: inserting a CDR record.
36.
Aug 26 16:07:35 DEBUG[2820]: cdr_mysql: SQL command as follows:
INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,
37.
billsec,disposition,amaflags,accountcode) VALUES ('2005-08-26
16:07:31','\"205Poly500\" <205>','205','*98','from-internal',
'SIP/205-6859','','ResetCDR','w',4
38.
,4,'ANSWERED',3,'')
39.
Aug 26 16:07:35 VERBOSE[2820]: -- Executing
NoCDR("SIP/205-6859", "") in new stack
40.
Aug 26 16:07:35 WARNING[2820]: CDR on channel 'SIP/205-6859' not
posted
41.
Aug 26 16:07:35 WARNING[2820]: CDR on channel 'SIP/205-6859' lacks end
42.
Aug 26 16:07:35 VERBOSE[2820]: -- Executing
Wait("SIP/205-6859", "5") in new stack
43.
Aug 26 16:07:35 VERBOSE[2820]: == Spawn extension
(macro-hangupcall, s, 3) exited non-zero on 'SIP/205-6859' in
macro 'ha
Re: [Asterisk-Users] All Page ??
Steve Maroney wrote: Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. I'm looking for an answer to this problem also. I am putting an Asterisk system into our new office. In our old office we used the old phone system to act as an intercom, you hit page all and your voice comes out of the speaker on several handsets throughout the office. This allows you to announce information or to the whole office, simply announcing to someones desk doesn't work since our people move around a lot and are not always at their desk. Anyway, I have some Polycom phones, and I have Autoanswer working with Asterisk, but which ever phone happens to answer the call first is the only one who's speaker my voice comes out of. Anyone have an answer to this problem? Thanks, Matt ___ 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] POLYCOM IP 500 Setup
Hello, I just wiped out my old asterisk install and installed Asterisk at Home. I was quickly able to get my Digium TDM422P working, 2 POTS lines, 2 phones. I also got X-Lite working as a SIP extension. I then tried to setup my Polycom IP 500, and this was not so easy... Using AMP I created SIP extension 205 to be used with my Polycom phone. I setup username = 205, secret = 123, context = from-internal. I setup my phone to have a static IP address, then pointed my web browser at it, to setup my phone. I setup Sip Conf with: Address = "IP of * server", Server1 = "IP of * Server" Under Registration, I setup: Identification: Address = "IP of * Server" , Auth User ID = 205, Auth Password = 123, Server1: Address = "IP of * server" Now with all of this setup this way, I don't see the phone registering iwth Asterisk, or even attempting to. What do I need to do at this point? Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newie Questions
Thanks for your repsonse, perhaps I mis-stated my situation. I have asterisk up and running with a TDM22B and have two analog phones working with two analog phone lines. What I can't seem to get started on is the setup of a SIP phone. I have looked at all the info on voip-info.org and it is somewhat helpful, but not enough to get it going. So any help would be appreciated. Also, is it generally accecpted that the Polycom phones are a good choice? Why might I choose something else? Can the Polycom phones be setup to work against a propritary phone system like the Nortel or Avaya? Thanks again, Matt Dean Collins wrote: Yes asterisk not only competes with avaya and Nortel but exceeds them once you know what you are doing. If you are only new to Asterisk there is now [EMAIL PROTECTED] http://asteriskathome.sourceforge.net don't be put off by the name - people run entire companies on this version) The [EMAIL PROTECTED] solution the easiest way to get started. It is an .iso cd that you burn, load into a suitable PC (I run mine on a P3-700) and this super smart scripting code automatically installs the following software; Asterisk (the open source switching software) AMP (an open source release of a gui configurator) they have their own separate sourceforge website https://sourceforge.net/projects/amportal FOP (a graphical web page for transferring calls, monitoring who is online etc) http://www.asternic.org Web meetme (a graphical web page for monitoring and controlling conference calls) Check out www.voip-info.org for information about configuring your Polycom Welcome to the family. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor Sent: Friday, 10 June 2005 5:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newie Questions Hello, I'm new to asterisk. My company is opening a new office and I'm seriously considering using Asterisk for the phone system. A couple of questions: How does Asterisk compete with the Avaya IP Office or the Nortel BCM systems? I have purchased a Polycom 500 phone but I'm having trouble getting it setup and talking to Asterisk. Is there somewhere that has SIP phone setup A-Z for beginners? All the documentation I have seen assumes you know more than I know at this point. I'm sure I'll have lots more questions, but that will do for now. Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newie Questions
Hello, I'm new to asterisk. My company is opening a new office and I'm seriously considering using Asterisk for the phone system. A couple of questions: How does Asterisk compete with the Avaya IP Office or the Nortel BCM systems? I have purchased a Polycom 500 phone but I'm having trouble getting it setup and talking to Asterisk. Is there somewhere that has SIP phone setup A-Z for beginners? All the documentation I have seen assumes you know more than I know at this point. I'm sure I'll have lots more questions, but that will do for now. Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
