similar to: Need help for transfer

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.

I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the "Forward" softkey just before she enters the "Page All" keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have

Situation: * New Install of Asterisk * 7960 w/ SIP 7.4 Image * 7912 w/ SIP040406A * 3 Lines Defined on the 7960 (5104,3100,2100) Questions (configs are below): * Why won't the MWI light on the Cisco? I've tried: * mailbox=2100 * mailbox=2100@default * mailbox=2100@wvlandsales-voicemail * Does anything look goofy overall? :-) Thanks, George Sip.conf [2100]

I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't

Seeking recommendations on handsets for use with Asterisk. I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942. Anyone using either one of these with comments on them? Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted

Hello list, I am doing some work for a non profit group. As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Hi all, I'm testing Linksys SPA922 phone and I have strange issue. when call is finished on the phone I see "CallEnded" and normal silence for cca. 5 seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on other phones I have tried (Cisco 7940, grandstream, XLite,... ) and I have to manually hangup handset to finish a call. Is this normal behavior

Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in "general" that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'd like to survey those on-list who actually use wifi SIP handsets. What type of wifi access point do you use? Are you happy with it? I presently use some older Linksys WAP54G APs. I'd like to replace these but in doing so I'd like to be moving in a VOIP friendly direction. I've yet to find a handset that I'd buy in quantity, but my last round of access points lasted >4

I have an asterisk server at home. I'm looking to replace my internal phones with sip cordless (DECT) phones. I'm now looking at the Siemens A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base ($80) and DP710 handset ($45). The Siemens has a feature were I can also use a PSTN landline, but I not sure that's a great benefit. Has anybody tried these phones? I

Has anybody else experienced problems with the Linksys PAP2-NA's? I've now had two of them fail unexpectedly, with no apparent rhyme or reason, having gone into a RED power LED, with a solid blue ethernet LED. No response from the device either on the network or from the phone.... To make matters even crazier, the one that now failed was the one I received as a replacement for the

I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The "problem" with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries.

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts

Mojo with Horan & Company, LLC wrote: > And it makes *clear* calls assuming you're within allowable range. > Speakerphone seems to work well too. I meant to mention that the DTMF tones and dialtone sound like they're played at such a high volume that they clip through the handset's speaker. DTMF is rfc2833, so what I'm hearing through the handset isn't affecting

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind

Is there a way to configure the linksys pap2 to automatically connect to asterisk on lifting the handset (presumably into the 's' state)? Asterisk would then be listening for DTMF tones to figure out what to do rather than having to put a dial plan into each pap2. I think the pap2 is pretty much the same inside as a few of the sipura boxes so the same thing might work if anyone knows...