similar to: Anybody having problems using sellvoip?

Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Thanks, SG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070325/1f30a3d3/attachment.htm

Has anyone been successful in getting a refund from SellVoip when you've cancelled service? Tom Lynn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070316/052ec182/attachment.htm

Try DIDx.net, I would not say they're best but at least they willing to help you when there is problem and they have a large pool of numbers. -------------- Original message -------------- From: "Salvatore Giudice" <Salvatore.Giudice@VoIPSecurityTraining.com> > I have transitioned to other DID's. I think that company is out of business. > > Sellvoip is best

hi guy, i have a problem, i have an sellvoip account and i want configure asterisk for outbound calls. this is my sip.conf register => XXXXX0000000000:PassWord@70.42.34.200 ; this is one of the sellvoip server [sellvoip_out] type=friend secret=PassWord username=XXXXXX0000000000 host=70.42.34.200 dtmfmode=rfc2833 context=testing disallow=all allow=ulaw extensions.conf this is a semplified

I was quite surprised to find a message in my in box from SellVOIP a day or two ago. It indicated I was running out of credit which was a surprise as I thought they'd gone under a large number of months back. So I ran upstairs, added their entry back to sip.conf, uncommented a couple of lines in extensions.conf and I'm again using sellvoip to make outgoing calls. The reason I was

Are you looking for local or Long distance? What will these T1s primarily be used for(inbound/outbound, domestic, local, long distance, international) How important are per minute rates to you? how many minutes do you expect to use per month? We are in Tampa Florida and have 15 T1s from several different providers so I may be able to refer you to one if it's a match to what you're

Hi: I'm trying to plot dates on the x-axis of a code, but the legend is not being displayed. I receive the following error: Error in match.arg(x, c("bottomright", "bottom", "bottomleft", "left", : 'arg' should be one of bottomright, bottom, bottomleft, left, topleft, top, topright, right, center In addition: Warning message: longer

This page http://www.freecall.com/en/index.html is advertising free calls to: Argentina, Australia, Austria, Belgium, Canada, Czech Republic, Denmark, France, Germany, Hong Kong (+mobile), Hungary, Ireland, Italy, Luxembourg, Malaysia, Monaco, Netherlands, New Zealand, Norway, Panama, Poland, Portugal, Puerto Rico (+mobile), Russian Federation, Singapore, Slovenia, South Korea, Spain, Sweden,

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly

(Apologies for cross-posting this job announcement to those on both R-help and S-news.) 2003 SUMMER INTERNSHIP POSITION Domestic United States residents / visitors only. Merck Research Laboratories Rahway, NJ, USA (approx. 30 miles from New York City) 1 position, Ph.D. or Masters student Deadline for Applications: February 14, 2003. Brief Description: Help provide data analytic &

Our company can offer VoIP to premises and domestic users and bill the premises as a whole. We need something to enable the hotel owner to bill each guest in a hotel in real time. What solutions do exist presently? (PS: Our radius (and every telephony equipment outside the hotel) does not recognise which room in the hotel initiated the international (VoIP) call, so that's the main problem

Hi, whatever dialplan I'm using for outgoing calls via PRI (Digium card, chan_zap), the callerid when receiving calls has no leading zeros, which are normally used to distinguish local, national and international calls in Europe. The number has always the area code in front, but the country code only for foreign calls. Now I'm looking for any mean to decide, whether the received

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

There was some discussion recently about the Solaris /dev/random support that can be downloaded from Sun's patch archive as part of a patch to the Sun Web Server 1.0 product. The SUNWski package is the interesting bit that purports to provide /dev/random. It was noted that domestic and international versions of the patch existed and that only the international (no encryption) version was

(Seems to me that Icecast folks would be particularly concerned about this. Please consider the following, lend your signatures, and also *send it on* to appropriate interested parties. If you are a blogger or know clueful bloggers, please try to have it posted in a highly visible forum. -- Seth Johnson) Hello folks, Please review the important joint statement below, related to the WIPO

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw