Re: [Asterisk-Users] Configuring Polycom 501 IP phones via the console
In my experience, this can be pretty cumbersome. I could be wrong but I think the reason I stopped doing it was that the phone would restart when you applied ANY changes, and you'd have to wait like 90 seconds or more to be able to re-access the phone via http. Moj Avi Miller wrote: Stephen Bosch wrote: All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. The console is very tedious. Why not use the web interface instead? Let the phone get an IP address via DHCP and then point a web browser at the phone. :) Much easier to navigate/configure. Password is the same as the advanced password on the phone itself. cYa, Avi -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring Polycom 501 IP phones via the console
Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks, -Stephen- ___ --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] Configuring Polycom 501 IP phones via the console
Hi Stephen, I use the 601's but I don't think they are THAT much different that this information won't be helpful or get you in the right direction. What is your network setup like? Are you using NAT or does the phone have a public IP address? Also are you seeing any errors on the CLI of asterisk? I know you said your configurations are local, but are you using a bootserver (which can be local) to grab the files? Things I would check if you are using NAT (I think 2-5 need to be done in the web interface): 1. Make sure your SIP.conf file is configured to use NAT and give it a port to signal on, say 1 for example (which I will use below to, but change to better fit what you would like). 2. Assign the phone an internal address, add port pass thrus for UDP packets 1-100050 (I think should be enough) for that IP. 3. Assign RTP port range to start at 10001 4. Make sure you have a NAT address listed in the phone and you have the signaling port set to 1 and Media start port at 10001. 5. Also if you are using a DNS name for the server (such as server-1.whateva.com) I use TCPperferred for DNS lookups. If you are not using NAT, it pretty much should work out of the box provided it knows where the server is going. Of course, make sure the SIP username and password are correct. I personally used a bootserver and manually changed my configuration files and I got my 601's working in no time. Hopefully something in here will help. Kevin Stephen Bosch wrote: Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks, -Stephen- ___ --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] Configuring Polycom 501 IP phones via the console
Hi, Kevin: Kevin Smith wrote: Hi Stephen, I use the 601's but I don't think they are THAT much different that this information won't be helpful or get you in the right direction. What is your network setup like? Are you using NAT or does the phone have a public IP address? Also are you seeing any errors on the CLI of asterisk? I know you said your configurations are local, but are you using a bootserver (which can be local) to grab the files? No NAT. This is just one Polycom 501 that is dialing out through an Asterisk server with a TDM-400 card in it. I'm not using a bootserver; I figured that with one phone, I ought to be able to just do it locally on the phone. The impression I am getting is that Polycom really doesn't want people configuring the phones that way. The Admin guide contains slightly more than *no* information on how to do that. It just seems like I should be able to enter a few things on the on the phone console and have it working, then fine tune things for larger deployments later. I just want to see the thing work first. Things I would check if you are using NAT (I think 2-5 need to be done in the web interface): 1. Make sure your SIP.conf file is configured to use NAT and give it a port to signal on, say 1 for example (which I will use below to, but change to better fit what you would like). 2. Assign the phone an internal address, add port pass thrus for UDP packets 1-100050 (I think should be enough) for that IP. 3. Assign RTP port range to start at 10001 4. Make sure you have a NAT address listed in the phone and you have the signaling port set to 1 and Media start port at 10001. 5. Also if you are using a DNS name for the server (such as server-1.whateva.com) I use TCPperferred for DNS lookups. If you are not using NAT, it pretty much should work out of the box provided it knows where the server is going. Of course, make sure the SIP username and password are correct. That's the trouble. So many places to configure! Example from the phone console: Menu | Settings | Advanced | enter password | Admin Settings | SIP Configuration Now you see a list of parameters: Server: ... Outbound Pro... ... [Outbound Proxy] RFC2543 Hold: No Calls Per Line K... [Calls per line key] Line 1: ... Line 2: ... Line 3: ... (Only one line configured for the Polycom in sip.conf, like so: [general] context=default srvlookup=yes [polycom] type=friend secret=welcome qualify=500 ;qualify peer is no more than 500 ms away nat=no ;this phone is not natted host=dynamic;this device registers with us canreinvite=no ;Asterisk by default tries to redirect context=internal;the internal context controls what we can do I've tested the installation with a softphone, and it works.) Here's one source of confusion -- The parameters in the Server: ... category are Address: [this is supposed to be the DNS or IP address of the SIP server] Port: 5060 DNS Lookup: UDP only [I set this to UDP only because the internal DNS server we're using here only does UDP] Register: Yes Now I have to set up the lines, so I go back up a level and down into Line 1: ... where I see Display Name: [don't know what this is for] Address: [what goes here? SIP server address again?] Label: [and here?] Type: Private [the other option is Shared] Third Party Name: [and what's this?] Auth User ID: polycom [here's where I assumed I had to put the extension name] Auth Password: [here's where I put the password welcome] Num Line Keys: [left this blank] Calls Per Line Key: [left this blank] After making those changes, I restart the phone. With Asterisk running verbosely, I never actually see the Polycom register. Not surprisingly, I can't make any calls at all. The phone is getting network information via DHCP. It does get an IP address and even configures the DNS right. (did you use the Polycom SIP admin guide to figure out how to set up your 601?) -Stephen- ___ --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] Configuring Polycom 501 IP phones via the console
Stephen Bosch wrote: That's the trouble. So many places to configure! snip Here's one source of confusion -- The parameters in the Server: ... category are Address: [this is supposed to be the DNS or IP address of the SIP server] Port: 5060 DNS Lookup: UDP only [I set this to UDP only because the internal DNS server we're using here only does UDP] Register: Yes Now I have to set up the lines, so I go back up a level and down into Line 1: ... where I see Display Name: [don't know what this is for] Address: [what goes here? SIP server address again?] Okay: I believe I have figured it out through trial and error. This Address field needs to contain the extension's address in the format: extension name@FQDN of SIP server So, in my case: [EMAIL PROTECTED] If I leave the address blank, the phone never registers. If I put the address in as shown above, it works! (I have been working on this for 2 days, so you can imagine I'm pretty pleased.) This information really belongs in the wiki, so perhaps I'll register and update the pages. I hope having this solution in the archives will help some other lost souls, too. I'm really frustrated that Polycom's documentation isn't clearer, though... -Stephen- ___ --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] Configuring Polycom 501 IP phones via the console
Hi Stephen, Sorry if the e-mail is a bit choppy but I figured it would be best to cut/paste answers in. Now again, I am using the 601's so things may be a little different, but for the most part should be similar. No NAT. This is just one Polycom 501 that is dialing out through an Asterisk server with a TDM-400 card in it. I'm not using a bootserver; I figured that with one phone, I ought to be able to just do it locally on the phone. The impression I am getting is that Polycom really doesn't want people configuring the phones that way. The Admin guide contains slightly more than *no* information on how to do that. It just seems like I should be able to enter a few things on the on the phone console and have it working, then fine tune things for larger deployments later. I just want to see the thing work first. I wonder if you are looking at a different guide. The Administrator guide I have (in Section 2.2.2) has a whole list of advantages for using a bootserver. If you are going to use FTP, then you need to make sure the phone has the proper information to access, same with HTTP. Then you just need the proper files up on the location. True, for 1 phone it isn't needed, but I am managing about 20 phones (some in different states and soon more) so it is very handy to have. That's the trouble. So many places to configure! Yes, I know, it took me about two days to get things finally sorted out, but once you get there...you will be like DUH! (Only one line configured for the Polycom in sip.conf, like so: [general] context=default srvlookup=yes [polycom] type=friend secret=welcome qualify=500 ;qualify peer is no more than 500 ms away nat=no ;this phone is not natted host=dynamic;this device registers with us canreinvite=no ;Asterisk by default tries to redirect context=internal;the internal context controls what we can do Okay, above looks fine. Now here may be some confusion. The sip entry isn't for a line...it is just a registration for Asterisk. The 601 for example, one key (which you will see later) can handle 24 calls (which is its max), The 501 can handle 3. But this just verifies the phone has access to the server, the context it belongs to, etc, the number of lines it can use is based on the phone and the available channels on Asterisk. Address: [this is supposed to be the DNS or IP address of the SIP server] Port: 5060 DNS Lookup: UDP only [I set this to UDP only because the internal DNS server we're using here only does UDP] Register: Yes Address is the address of the SIP server. Port: 5060 which is default For DNS, if you can only use UDP that is fine., and of course you want the phone to register. Now I have to set up the lines, so I go back up a level and down into Line 1: ... where I see Display Name: [don't know what this is for] Display name, is caller ID basically. If you have support for caller ID name, that is what it is. I do fill it in, like for example my company's name is on my phone config, but I don't see any reason why you can't leave it blank. I was thinking ahead for if/when we do SS7 or something the name will show up. Address: [what goes here? SIP server address again?] This is a little confusing, but this is the number or extension. For example, a phone number. You also can dial Internet addresses so that is why it is called an address. I believe this is also used later... but for now, I would set this to your extension, even if it isn't used, it is there for when it is. Label: [and here?] One the phone, next to the line keys, this will be the label..such as Line 1, or My Phone, it will show up there. Type: Private [the other option is Shared] I leave it at Private Third Party Name: [and what's this?] According to Polycom, this field must match the registration address value of the other registration which makes up the bridge line...what did I do with it? I left it blank. Auth User ID: polycom [here's where I assumed I had to put the extension name] Yes, however, again I use our phone numbers both in address and here...why? Because it was much easier to code in my opinion. I think if you leave this blank, it will use the address, but I'm not sure, which is why I matched it. Since polycom is your name in SIP you will want that there. Auth Password: [here's where I put the password welcome] Yes Num Line Keys: [left this blank] Calls Per Line Key: [left this blank] Here is what I was talking about earlier. Num Line Keys, is how many keys for numbers. For example, if you set it to 2. On the right of the LCD screen you will see a graphic of a phone in spots 1 and 2 and your contacts (if any) would follow. For starters I would set both to 1. Now, if you change calls per line key to 2, then it is like you have call waiting. You will be on a call and you will hear a beep and see on the phone someone else is calling. After
RE: [Asterisk-Users] Configuring Polycom 501 IP phones via the console
I too had the same problems. If you find out the best way for this let me know! Thanks Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Sunday, June 04, 2006 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Configuring Polycom 501 IP phones via the console Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks, -Stephen- ___ --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] Configuring Polycom 501 IP phones via the console
Stephen Bosch wrote: All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. The console is very tedious. Why not use the web interface instead? Let the phone get an IP address via DHCP and then point a web browser at the phone. :) Much easier to navigate/configure. Password is the same as the advanced password on the phone itself. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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
