similar to: Leadtek BVA8051 / Sipphone.com CallInOne with Asterisk?

Hello, can anyone comment on how one could use SIPphone's $89 All-in-One adapter with Asterisk? Sounds to me like it should work as both a FXO and FXS. It would be a cheap way of getting started with Asterisk and PSTN. Any comments on the SIPphone FX200? Any comments on SIPphone in general? Thank you for your help

Hi, A friend has a few hundred deployed LeadTek BVA8055's and needs to bulk re-provision them. There isn't much documentation on the web. Anyone have documentation explaining the LeadTek provisioning process and the provisioning file format? -- Eric Chamberlain

**** DRAFT **** DRAFT **** DRAFT **** DRAFT **** I've been using the DTA310 from Packet8.net for a couple of weeks. The DTA310 is about $130 without the Packet8.net VoIP service. It only supports SIP. On the back of the DTA310 is a power connector (power supply is provided with the product), a 10/100 Ethernet port, an FXS port, and a reset button. The front of the device has LEDs for

I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware version (Application Code Version: DTA version 1.0 US (8x8 001111)) onto it via TFTP, so I could access the SIP configuration. Under the SIP config, I put the IP of my * system, the 5060 port, and for Domain Name, I put default (is that right?). I checked off the Send Registration Request box. Dial Plan I left at the default,

When attempting to call one of the example numbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

Hello All. I started setting up my Asterisk system yesterday and everything was going well, i have registered with sipphone.com and set-up my Asterisk system to register with sipphone per the sip.conf file below. It was registered perfectly but I could not receive calls so I added in the line "insecure-very" and I then used the Washington DC access number to test and the phone

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing

Has anyone else had problems registering with Sipphone over the last few weeks? Previously, this had worked fine. I contacted Sipphone technical support, but they're not much help. register => 17471234567:password@northamerica.sipphone.com/123

Come to think of it since the DTA310 uses DNS to find the SIP server, you could setup a DNS cache and override the DNS entry for what packet8 uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your own SIP server? Kind of a hack but it should work as long as it's running on port 15062. I am very new to this so I don't know if there's a port standard for SIP

Hi all. I found a connection error from sipphone.com. It seems 'realm based authentication' by sipphone.com. any ideas? Regards. mack

Should SipPhone support ISN routing for their 747 ITAD? Cast a vote: http://forums.gizmo5.com/viewtopic.php?t=10197 Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD #1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever (Karl tips his hat to Ward Mundy) and it's also really, really funny.

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

All of a sudden, my account doesn't appear to work, or even perhaps exist with SIPPhone. Is anyone else having trouble?

Any sip softphone will work.? Linphone is free. I have tested many . All works well with audio.? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131102/21e533b0/attachment.html>

[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) &

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File