search for: marlowe

I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show. It's simple while the phone is ringing that it doesn't display. I mean I doubt the polycom is malfunctioning, that's why I t...

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 =>

Has anyone been able to get * working with VoicePulse OpenAccess (SIP not IAX). I have found a ton of information about VoicePulse Connect but very little on the proper * settings for OpenAccess. Tried contacting VP with no response. If anyone has this working, can they share their extensions.conf and sip.conf files? Better yet, if it could be posted on the Wiki. Keith

Is the 7905G Firmware the same as the 7960 firmware? -- MBM

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,

...method ''exists?'' to return true when its called twice. Again, it worked fine in all rails versions up to 2.0.2, and only fails with rails 2.0.2... Is it me? -- Virtually, Ned Wolpert http://www.codeheadsystems.com/blog/ "Settle thy studies, Faustus, and begin..." --Marlowe

...m likely making my test case too complicated if I have to do this, but we found an esoteric bug on a short-lived object and having a test to check for this specific case would help make sure that bug stays away.) -- Virtually, Ned Wolpert "Settle thy studies, Faustus, and begin..." --Marlowe Discere docendo...

I wish 'A(x)' was available with AgentCallBackLogin!! :( -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo Brancaleoni Sent: Tuesday, February 03, 2004 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation show application

Those of you who track applications of R may be interested in the following: "The purpose of this paper is then to apply modern text analysis techniques using the R statistical packege [sic] to compare the works attributed to Shakespeare to those of leading alternate candidates such as Sir Frances Bacon, Christopher Marlow, and Edward de Vere...".

I'v got the following scenario: Two clients (ohphone) are calling (one at a time) the host with asterisk, which then connects to the SIP client. One of these hosts let's asterisk crash with a segmentation fault (i can provide the core file, if needed) in the second, the SIP client accepts the call. However .. if that client get's to the voicemail instead, because the SIP client is

Hello all, I have been trying set up a linux box as a print server for our mixed windows network. It has samba working - everybody can see it & a shared directory. I installed cups & that too seems to be fine - printing test pages from the web interface works perfectly. But i can't get them to talk. I think this is mostly as i can't find a decent howto & so am poking

We''re encountering a failure with Mocha 0.5.6. We had this expectation: game_version.expects(:attributes=).with(:game_file => kind_of(GameFile), :game_id => @game.id) This expectation was passing with 0.5.5, but fails with 0.5.6. I added this test to parameter_matcher_acceptance_test.rb, which passes in 0.5.5 and fails in 0.5.6 def test_should_match_nested_parameters

Hi, Does anyone know whether subsections are specific to the gnu assembler or whether they are supported by other assemblers, such as masm? Or put another way, will this limit the assembly output to the gnu toolchain? Cheers, Sam -----Original Message----- From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] On Behalf Of David Terei Sent: 15 June 2010 14:18 To: Andrew

I''m a newbie. My first app is using a table called cases. With just "scaffold :case" I get the basic list to show up. I just did a "generate scaffold case case" (I think that is correct). Now when I try to list my cases, I''m getting a syntax error on all of the view source code that references "case". ie. <% for case in @cases %> Thanks

I've got Zaphfc working running Asterisk v. 0.7.2 Then I have tried with Asterisk V. 1.0 and the latest from CVS - with no succes. Has anybody got zaphfc working with newer version than 0.7.2 ? NR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040616/32481fca/attachment.htm

I recently got a 7905G w/ Sip software preloaded. I got it working w/ asterisk with no problem setting it up through the phone. I am now trying to make it download the config file from the tftp server. I have set all of the options in the file and the file is definately named correctly. But the phone is simply not processing the config file for some reason. Two commands Im trying to get

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when 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

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