search for: ip450

Only slightly asterisk related I suppose, but hoping someone has attempted this... I have an old installation with a bunch of IP501s, and one died. I replaced it with an IP450, and the user sorely misses his DND button. I hated those DND buttons anyway, as I couldn't control them centrally. I'd *like* to program one of his softkeys to send a *XX sequence to do DND on the asterisk-side instead of on the phone. Has anyone remapped the softkeys on an IP450 thi...

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a cal...

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

...o Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.1.70 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Transport : UDP Def. Username: Barry's IP450 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No 100 on REG : No Status : OK (14 ms) Useragent : PolycomSoundPointIP-SPIP_450-UA/3.1.3.0439 Reg. Contact : sip:6668 at 192.168.1.70 Qualify Freq : 60000 ms Sess-Timers : Accept...

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 100...

...most part reasonably-priced. I'll let Michael discuss this with you. He was recently lambasted on Om Malik's blog for calling SIP for Skype "vaporware". I didn't see his comment as meaning that, but we'll find out more tomorrow. Also it's time to give away the Polycom ip450 from e4strategies so we will be doing that on tomorrow's call as well. You will need to be on IRC and on the call to be eligible and be registered with Talkshoe in order to have a PIN: IRC: #voip-users-conference on Freenode.net or via the web: http://tr.im/vucirc SIP: 7463#22622#${YOUR_PIN_H...

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...e in ours), to provide phone service during this "mini disaster" - with a Wifi connection to the HQ building and VoIP, we had full services within a day or so, rather than weeks if we had waited for the telephone company. We're now moving the IT department over to the system (Polycom IP450s) permanently, as it has worked so well with the Cisco ATAs. Next we plan on replacing the voicemail/auto-attendant box on the Toshiba with Asterisk. Following that, the next step will be to eliminate the incoming POTS lines into the Toshiba and replace them with an ISDN PRI line going into the As...