similar to: Sellvoip configuration....Please Help!!!!

hi guys i have created a plugin for jquery for asterisk ajax interfacement. the interfacement work with ajam and on firefox work very well, the problem is with IE :-( an example: the url is: asterisk/mxml i want login on manager system and the string command is: action=login&username=myusername&secret=mysecret I have tested with firefoz and i receive the correct XML response, the

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

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

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

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

Can someone get this guy off the lists? ---------- Forwarded message ---------- From: postmaster@prebit.net <postmaster@prebit.net> Date: Nov 1, 2006 3:22 PM Subject: Benachrichtung zum =?unicode-1-1-utf-7?Q?+ANw-bermittlungsstatus (Fehlgeschlagen)?= To: joakimsen@gmail.com Dies ist eine automatisch erstellte Benachrichtigung +APw-ber den Zustellstatus. +ANw-bermittlung an folgende

Skipped content of type multipart/alternative-------------- next part -------------- An embedded message was scrubbed... From: "Sales Support" <sales@sellvoip.net> Subject: {100-1287} RE: DID"s Date: Tue, 18 Oct 2005 11:04:09 GMT Size: 1774 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20051018/7221e0af/attachment.eml

International calls (Germany) haven't completed since around 3/1. Domestic works. Is it just me? I'm getting 503 responses. Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070306/02b028cf/attachment.htm

On Thu, 2017-01-05 at 10:45 +0100, Stefan G. Weichinger via samba wrote: > Am 2017-01-05 um 00:28 schrieb Andrew Bartlett: > > > > > > > > > In Debian we re-bundled heimdal.  Samba is only known to work and > > > is > > > only tested with the bundled copy.  The semi-private interfaces > > > we > > > use > > > with the

Hi I looking a good IAX service for a *emerging * voip provider. Better with a test account to try. Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte t?cnico ISPs Jabber ID: rpereyra@lugmen.org.ar For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 -------------- next part

Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

Hi, I would like some suggestions. I''m wondering if it''s better to upload a file directly into a database or to put the file into the server and keep the path in the database? Thanks, Eric -- Posted via http://www.ruby-forum.com/.

I have a single DID number from sellvoip.net. This works well and has been mostly reliable (HEH). I am wondering if caller id info is likely be provided for calls to my DID #? If so, should they show up in the CLI(they don't)? Or how do I go about looking at them (the caller ID infos that is)? Thanks, Marty

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

I am trying to use dovecot without ssl in centos 7. It works with imap but I am not able to make it work with pop3. Pop3 only works with ssl=yes. I can use dovecot in centos 4 ok.? I get the same error with thunderbird and outlook. But for some reason it works when I telnet to port 110 and do retr 1. I was able to work pam and shadow passwords. this is the output that does not work with tls

On Thu, 23 Mar 2006, syslinux-bounces at zytor.com wrote: > Your mail to 'SYSLINUX' with the subject > > Re: [syslinux] Syslinux keyboard mappings > > Is being held until the list moderator can review it for approval. > > The reason it is being held: > > Message has a suspicious header > > Either the message will get posted to the list, or you

On Thu, 2017-01-05 at 12:26 +1300, Andrew Bartlett wrote: > On Thu, 2017-01-05 at 00:07 +0100, Stefan G. Weichinger via samba > wrote: > > > > Am 2017-01-04 um 23:49 schrieb Stefan G. Weichinger via samba: > > > > > > > > > > > on my way to 4.5.3 now > > > > compiling samba-4.5.3 on gentoo *failed* right now > > > >

I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? THanks, JErry ---------------- <Date>03/06/2006

Sometimes the direction of UI development in gnome really angers me. For example, when selecting a font for the gedit text editor - there is no way to ask it to only show monospace fonts. It's a fricken text editor, that should be the default - meaning you have to do something special to get fonts shown that aren't monospace. Seriously, who is in charge with the UI design in gnome?