I asked this on the Asterisk forum as well as on a Polycom forum, but didn’t get an answer. Here’s my problem…I have an extension, 1367. The user wanted it to be 1372. So I changed the name under SIP configuration, and under the phone’s config. files I changed the information in there as well. When the phone tries to register I got an “username/authname invalid”.
I checked my sip.conf and the secret is ts1372. I checked the phone’s config. file and the auth username=1372. The password is ts1372. I checked the phones sip configuration files, and that information is in there. I don’t know what else to do. Any ideas?
The answer is pretty clear. Some where in the extension settings and/or the phone config there is a username/authname configuration issue. The best bet is start by deleting the extension as there is really not a way to rename the extension number.
ok I’m confused, you are asking for help on the FreePBX site for a phone system that does not use FreePBX?
You did not state that in your posting, we here at least assume that you are using FreePBX and try and help but your chances of getting help here are close to slim and none.
How about being forth coming and at least post all the details upfront and maybe somebody who might have knowledge of that system might have a answer but if you don’t way who/what it is I know for a fact you’ll never get a correct answer.
This forum is filled with many great minds but none of us are mind readers.
How about reading the first post where it says I asked this question on a Polycom forum, and on an Asterisk forum but got no help. If it was FreePBX related I think I would’ve asked here first. What do you think?
My hunch was that someone who goes to this forum has dealt with a Polycom independent of FreePBX, DRUID, etc.
Yes you said that you asked on other forums, but that didn’t mean that you were not using FreePBX. On IRC many times people go and ask questions in the #asterisk forum before asking on #FreePBX and use FreePBX… So you were not being clear and still are not.
For anybody who might be able to help how about saying what you are using so that if somebody knows they can help?
This a pure asterisk setup? if so what version, if it’s something else like freeswitch say so. People don’t get banned here for using other things.
But being forth coming, clear and upfront with all the details goes a long ways in getting the correct help. No instead of doing it at first they have to read down to the nth posting for info that should have been in the first one (yes you can re-edit the first posting to put it there). A lack of details is what kills people here from getting support. If somebody willing to help has to guess or make assumptions then when it’s wrong people get upset. So how about the extension settings and the lines you changed in your polycom setup to try and get the new extension number to work?
As for Polycom setup’s I have one, and I’ll be the first to admit I’m no expert. To be quite frank compared to the Aastra and Cisco phones I’ve setup. Polycom is a royal pain in the ass with it’s layered files, etc. I seem to have to edit/configure many more lines and files for a Polycom setup then I do for either of those.
You’re asking for specific answers to your problem, yet you’ve only given vague details as to your setup.
What type of Polycom are you using ? SIP300? IP501 ? Soundstation2W ?
Have you changed the user authentication in the Polycom phones’ web page ?
Was this phone auto provisioned ?
What happens in Asterisk (asterisk -r) when you try registering the phone, what errors come up ?
How about reading the first post where it says I asked this question on a Polycom forum, and on an Asterisk forum but got no help. If it was FreePBX related I think I would’ve asked here first. What do you think?
If you’re wanting some free assistance, how about you you try being a bit more polite ? What do you think ?
There’s dozens of things, it could be, but if you’re wanting the right answers, you need to help us, to help you.
So, post what version Asterisk you’re using, the Polycom phone model, the capture file, etc.
Actually, politeness has nothing to do with it. It’s just understanding on both parts. I posted what I thought was a fairly basic question that anyone would know who has worked with this stuff. So I kept details out that I thought you guys wouldn’t need. (None of this stuf is obvious to me!)
I’m using a Polycom 601. How could I quickly tell the Asterisk version? It’s for a remote client. The phone used to be a functioning extension of 1367. The client wants it to be 1372. So I changed the information under SIP configuration on the phone. Then I got all the .cfg files associated with the phone, and made sure that they now read extension 1372.
I made sure that username and password in the config. files reflected what I put on the phone. (I edited the config. files using nano). Asterisk “sees” the phone, but it’s telling me that the phone has an “authname/username” mismatch.
Forgive my rudeness, but I have a lot of tech issues to deal with. The main tech here is supposed to train me on this stuff, but he doesn’t want to. Since I need to keep my job I have to figure out what I can. So needless to say I get very frustrated and forget to post any information that would help.
As for asterisk version go into the asterisk cli (command to get there is asterisk -r) and it will show you that version when it starts the CLI. Most people (yes there are those asterisk purist that do everything the hard way) use some form of a Gui like FreePBX, asterisk now (now moving to FreePBX), trixbox pro, etc so knowing if there is one and what it is is very critical. Mainly most GUI frontends expect only certian files to be edited and without knowing the GUI just editing the files can be VERY dangerious.
To see what if any gui is on the server open a web browser and point it at the phone server and see what comes up.
But the lack of details is what is killing you here in getting help. It’s like saying my car will not start please help me fix it. When I do that we’ll have to go through the auto mechanics standard 20 dumb questions guide to begin to get to a valid starting point Or I could be smart and provide all that detail up front and not wait to be asked for it.
As for the polycom side of things can you provide your .cfg config file file for the phone for a start, and the customized file you edited for the extension info also (don’t know what name your guy called it in your setup) mine are xnnn.cfg where nnn is the extension number.
It’s Asterisk version 1.2.24. When I go to the IP it comes up as Druid, however, it wants me to install a license before it allows me to access anything. When I click for a trial license the Voiceroute site reads, “The page you have tried to access cannot be found.” I was told that we make changes/access the Asterix box via SSH. So that’s how I configure.
I’ll post the files but here’s probably a silly question…is there a way to copy and paste the file contents without copying everything on the screen to the clipboard? I’m using Putty. When I copy to clipboard it does EVERYTHING on the session not just what is highlighted.
I don’t know how Driud works it’s not one of the packages I’ve played with. But it being a GUI that would explain why nobody on the asterisk forums will answer as they don’t like GUI’s and want everybody to be a purist and hand edit the files.
I’ll also be of no help on Putty, that is one of many reasons why I don’t use it. When using windows I use Securecrt, have for years and love it.
Here’s what I think may be the problem…the client just told me that another phone they disconnected was extension 1372. When they reconnect it the phone comes up fine. I guess that Asterisk authenticates the original 1372, and refuses to authenticate the extension I changed to 1372?
Cipher ---- Just a suggestion here. If you have changed the config files and the phone does not reconfigure … then either your TFTP or FTP is not working correctly or your config files are incorrect. In which case you can log into the actual phone and make the changes …web to the IP address of your phone with default user name as Polycom and password is 456. However this may be a mute point per your last post indicating duplicate IP address or the same MAC address some where.
it seems that “cipher7836” is a complete newbie to the concept of a phone server (server) and a phone extension (client). And to make things even more interesting he may never heard of an asterisk…just a guess.
cipher7836, polycom phones can be used with various types of phone servers, specifically with sip based phone servers, asterisk is the most popular sip based phone server out there, chances are you may have googled and ended up with the named asterisk and you figured that asterisk and polycom’s are the only combination.
its a good possibility, that your specific phone system may have polycoms proprietary server, in which case, none of us here can be of assistance to you.
That was a pretty rude reply. I’m no expert at this but I just upgraded 19 Cisco phones to SIP, and configred them for a client yesterday. So far they work just fine. Sorry that I’m not as bright and experienced as you are.
As for everybody else that replied, thanks for the information and help. It was really appreciated.
It turned out that the FTP server I setup on an internal Windows box would not allow the phones to download the config. files, and the Asterisk server is in an entirely different state. I configured a soft phone with the same settings, and it worked fine. The client will be mailing me the phone since I’m not allowed to make any changes to their boxes over there to get the FTP/TFTP server to work.