similar to: Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?

I got most of the features of my phone working. Polycom TEch support refuses to help or even talk to me. So I'll have to ask here again. On incoming calls, only the NAME is displayed. I am trying to figure out how to get the NAME & NUMBER displayed. If anyone can help me do this it would be GREATLY appreciated. Thank you in advance

Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on

I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

In ipmid.cfg I have: <G3INTERCOM se.rt.10.name="G3INTERCOM" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.10.ringer="7"/> In sip.cfg I have: <alertInfo voIpProt.SIP.alertInfo.1.value="G3INTERCOM" voIpProt.SIP.alertInfo.1.class="10"/> I set up a test extension: exten =>

Lately I have been experiencing a lot of echo from my Polycom phones. Only I hear the echo and it's not on every call. I've researched it via google and the forums and every echo problem usually relates when it's using a Zap card and not an IAX provider. Can anyone give me some advice or where to look to help solve this echo problem? This never occurs on any of our other phones,

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

I've managed to use GhoustScript (gs) to take a postscript file and convert it to tiffg3, but I CANNOT seem to make it merge multiple files. Here is the output from tiffinfo on the file that SG generates: fteTYGeh2v.tif: TIFF Directory at offset 0x8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 1056 Resolution: 204, 96 pixels/inch Bits/Sample: 1

Folks- With Rails 2.0.2, I''m seeing mocha failures where they did not fail with Rails 2.0.1 test_set_to_process_fail_in_transaction(FileRecordTest): NoMethodError: undefined method `ord'' for 63:Fixnum /home/wolpert/svn/rubylms/vendor/plugins/mocha/lib/mocha/class_method.rb:45:in `hidden_method''

I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up <<< NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by

Folks- I just started using Mocha and I wanted to try something but I couldn''t get it to work. I''m coming from the Java world, so my approach may not be optimal for Ruby. Say I''m testing a.do_something() which calls b.do_other_thing() twice, but I really don''t want to change b.do_other_thing(), just ''expect'' it to be called twice. I know I

Just out of curiosity, What version of CVS and Polycom SIP software are you running happily? Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0? I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with poor results. Transferring did not work as expected. Using the # key to do blind transfers after a call was on hold did not work. Just curious. Thanks, - Brent

Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

On Fri, 2004-05-07 at 16:30, anthm@lists.digium.com wrote: > Update of /usr/cvsroot/asterisk/apps > In directory mongoose.digium.com:/tmp/cvs-serv17955/apps > added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call > +" 'D([digits])' -- Send DTMF digit string *after* called party has answered\n" > +"

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

Thanks, I was afraid to try and change gains that I didn't know what they did simply because I don't want to blow a speaker or something.. :) I'll try it today. The only thing I haven't figured out is how to set a default ringer in the configuration file, set my time to EST w/ Daylight Savings and when receiving incoming calls if it's possible to see NAME & NUMBER instead

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Can someone point me in a direction where I can acquire the Cisco 7960 SIP firmware? I just purchased two phones of ebay, but they are both in Call Manager mode. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030515/6265b777/attachment.htm

now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the "hands free" mode? Thanks

I've had an IP300 for a while now and it's been working fine. I just got an IP500 and when it connects to the FTP server it downloads the new bootrom and says error loading. The bootrom is fine and works on the 300... In addition, I downloaded a new copy to be sure and it still doesn't work. Can anyone give me some advice?