search for: onsip

Hello Alex, We have a customer looking to replicate OnSIP using OpenSER/Asterisk or FreeSwitch. Can you provide us a quote on the cost to completely replicate OnSIP? Thanks in advance, Ed Direct: 678.522.8511 Mail: edpimentl[at]gmail.com] Voip/IM: edpimentl [SKype | GoogleTalk ] -------------- next part -------------- An HTML attachment was scrubb...

...Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: talkshoe at vuc.onsip.com (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: 7463#22622#1 at proxy.ideasip.com (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us...

...able phones - Skype Outage - VoIP on mobile devices - or perhaps something more personal..... Come one, come all. Bring your story. Connect details at http://vuc.me Michael Graves -- Michael Graves mgraves<at>mstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgraves at mstvp.onsip.com skype mjgraves Twitter mjgraves

I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think).... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081205/fdd14af9/attachment.htm

I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

...rection. I've yet to find a handset that I'd buy in quantity, but my last round of access points lasted >4 years so changing these now will merits the voip consideration. Thanks, Michael -- Michael Graves mgraves<at>mstvp.com o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

...list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? I'm considering something like http://www.mobigater.com/index.php?p=5 Thanks, Michael -- Michael Graves mgraves<at>mstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

...not the major consideration as long as they're in >> the same range. (under $175) >> >> Michael >> -- >> Michael Graves >> mgraves<at>mstvp.com >> blog.mgraves.org >> o713-861-4005 >> c713-201-1262 >> sip:mjgraves at pixelpower.onsip.com >> skype mjgraves >> fwd 54245 >> >> >> >> > > > >The 430 has a larger footprint and feels to be a slightly heftier and >more business-like phone. I have very similar sound quality and results >with either phone. To me it would be a to...

Hi, As usual, you can get all the dial in information at http://VoipUsersConference.org IRC is on Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: talkshoe at vuc.onsip.com (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: 7463#22622#1 at proxy.ideasip.com (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us...

...esuming that the system is on a limited bandwidth connection is would start to prefer a compressed codec as the call volume increased? Perhaps shifting from G.711 to G.729? Michael -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgraves at mstvp.onsip.com skype mjgraves fwd 54245

To celebrate three years of the VoIP Users Conference, we're doing a 24-hour VoIP conference call today. Details are at http://voipathon.org IRC: #vuc on Freenode.net SIP: voipathon at vuc.onsip.com - Enter 22622# and your PIN# if you have no PIN you can listen using 1# iNum - +883 51007 039 9924 PSTN: +1 724 444 7444 again, 22622#1# or PIN# if you have one. Those of you in the Southern Hemi, here's your chance to stop by and say "hi". Look to your skies for a warning.....

...they have a bounty on offer for this if someone's interested in doing the work. Would the availability of the Firefox code make it easier to do an ActiveX implementation? Any takers? Michael -- Michael Graves mgraves<at>mstvp.com o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

Hi there, in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. Here's my overview and review so far... http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Cheers Al -- The way out is open! http://www.theopensourcerer.com

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

Michael Grave just posted a question about surround conferences. http://www.facebook.com/notes.php?id=564633430#/note.php?note_id=5009726 3908&id=564633430&index=0 I didn't see it posted on the ast-list, what do you think? Does something like this have potential? I'd love to listen in on one of these calls to see how it actually sounds if someone builds a trial

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:

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin