similar to: Polycom IP335

Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well.

Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live production. I cannot run valgrind and I do not have the right flags set in menuselect. I can however at the dead of the night run stress tests. I want to simulate x-amount of concurrent calls to both a dtmf dialplan, which is working, as well as MoH dialplan to see if this could be the cause of crashing. How

I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller name / number. Has anybody else noticed this with 3.3.1? I had thought with 3.2.4 it would automatically show call waiting name and number without pressing any keys. It could be

On Tue, 4 Jan 2011, Andy Graybeal wrote: >> The Polycom 321 has not been EOL'd and supports VLAN. It is, however, >> lacking a 2nd ethernet port if you were to go that route. >> >> -M >> > Thanks for the response Mark. I see the 331 has two ports and the same > features as the 321. > > I'm wondering what phone would be best being used as an

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

Hello. I have a dataset with about 500,000 observations, most of which are not unique. The first 10 observations look like 901000000000100000010100101011002 901101101110100000010100101011002 901000000000100000010100000001002 901000000000100000010101001011002 901000000000100000010101010011002 901000000000100000010100110101002 901000000000100000010100101011002 900000000000100000010010101011002

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

I trying to compile R-.1.7.1 under OS X. Configure worked fine; then make crashes with: /sw/lib/libg2c.a(err.o) definition of common _f__formatted (size 4) /sw/lib/libg2c.a(err.o) definition of common _f__hiwater (size 4) /sw/lib/libg2c.a(err.o) definition of common _f__putn (size 4) /sw/lib/libg2c.a(err.o) definition of common _f__reading (size 4) /sw/lib/libg2c.a(err.o) definition of common

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

Hey all, I tried to do some searching but I found snippets and I am having trouble putting it all together. I want to have an option off the IVR that plays back the announcement for the day. At the end of the message, I want the caller to get kicked back to the previous menu. The conditions are that I want the recorder to dial a feature code that prompts him to record the message. He

I'm curious as to what versions of everything you are using. Reason being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing it to SIP/5000-00000000". It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to SIP/801-0000000c" [1-1 being the span and channel

Please forgive me if this has been answered in another post. I''ve looked around but couldn''t find a solution. I bought the book "Agile Web Development with Rails" and think I''m going to really like it. I went through the first chapter with no problems. However, as I begin developing the database on page 57, I get a consitent error: Before updating scaffolding

Hi all, I am writing a custom backend. Doing more testing i notice that for some reason something like: int test(int x) { return x - 1; } is being turned into this IR: ; Function Attrs: nounwind define i32 @test(i32 %n) #0 { entry: %n.addr = alloca i32, align 4 store i32 %n, i32* %n.addr, align 4 %0 = load i32* %n.addr, align 4 %sub = sub nsw i32 %0, 1 ret i32 %sub } But finally in

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know