similar to: FXOs

Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can

I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

Sorry if I post in this forum, this may be not the right one, but I hope to find in here some experts which could help me out. I have in one location 8 extensions from a Panasonic PBX ?KX-TD1232? connected to FXO Ports on an MultiVoIP Gateway from Multitech. On the other location I have 8 SP5100 IP-Phones from Micronet as remote extensions. Between the locations I have a VPN.

Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter >> I am located in the UK and

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

Folks, I'm looking at buying an FXO card for my home Asterisk setup. The question I have is, should I got for a the Digium Wildcard FXO or settle for the winmodem that Digium sell? I have spoken to the UK distributor and they only sell the Wildcard. They said that the quality of the modem was questionable. Does anyone have any experience of this? Is the modem noticeably different

Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ?

I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-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

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

Hello, does this sound as a good combination, mini-itx board with Atom dual core 1.6ghz 2G ram and a sangoma USB? For a setup with PSTN for incoming and IAX2(alaw/gsm) for outgoing calls. - Would you say its a good choice from a hardware perspective? - Roughly how many concurrent calls would one of these be able to handle? Regards, Veselin K

I apoligize in advance for this newbie question on what I perceive as a mostly advanced level list... I did some searching, but would like some of your expert opinions. I'm building an asterisk server to be used in the home, both to learn, and as proof of concept of applying this solution in a home. To keep costs down, I'm considering one X100P ($25 ebay clone) card to connect the

Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

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

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

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

Would anyone care to offer opinions as to the FXO interface which sucks the least :) I have an application in which it appears I must route certain calls out an analog PSTN line. Presently, I am testing an SPA-3000, but I can't seem to get the echo heard on the IP end of the call down to a non-annoying level. Any suggestions welcomed :) -Dorn

Hi All, Has anyone on-list tried using a USB style adapter like the VTA-1000 to provide some form of gateway from Skype into *? If so, how well did it work? Would I need to combine the VTA-1000, which appears to be USB-to-RJ11 and a standard FXO port? Would it be possible to provide a more direct approach? Some of my associates use Skype, which I'd prefer to avoid. However a single

Hi, Have posted to this list a couple of times and have always received great responses and help. I have a basic * system setup Using 3 X100P cards with 6 Snom200 IP phones. It was a bit of a struggle getting everything up and running but have been pretty happy with the flexibility and ease of *. My major problem is one that has been discussed on this list many times before. The echo

Hello, I have received yet another new phone today, the ClipComm 101 (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html) I bought it for $165 directly from the Korean Manufacturer(No US distributer yet). Here are the features: - Built-in NAT functionality, you can switch from Hub to Nat, great for home DSL/Cable users - This includes some limited port forwarding