similar to: Opinions on the best wholesale origination/term providers

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

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

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

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

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

hi folks, Has anyone found a good (and, ideally, cheap -- we don't really want any per-minute charges) origination provider which can handle a moderate number of simultaneous incoming calls (to the same, single DID)? Many of the providers I've tried contacting either won't call me back, or want me to sign an NDA just to get a rate quote, or some other bullshit. Most of the

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

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

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

Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can.

Mike, Martin O'Shield here from WindyCitySDR. >From: mike castleman <mlc@democracynow.org> >Subject: [Asterisk-Users] origination providers >To: Asterisk Users Mailing List ><asterisk-users@lists.digium.com> >Message-ID: ><20050524190016.GG11096@pinetree.mlcastle.net> >Content-Type: text/plain; charset="us-ascii" >hi folks, >Has

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

Dear Asterisk Community, Does your company provide inbound 800# origination? If so, please read this message and e-mail us a quote for monthly co-lo hosting of our asterisk server and per-minute inbound 800# origination. The Prostate Cancer Research and Education Foundation (PC-REF) is a non-profit organization dedicated to helping prostate cancer sufferers and their loved ones. We have

Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten => *601,1,Answer exten => *601,n,Dial(IAX2/pbxnetwork/xxxxxx,30,m) exten => *601,n,Hangup exten => *602,1,Answer exten => *602,n,Dial(IAX2/pbxnetwork/xxxxxx,30) exten => *602,n,Hangup No I have a problem when

I'm looking for sip origination and termination companies. Anyone know of reliable ones? I am interested in wholesalers also. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090417/644bcbad/attachment.htm

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/.

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 > > > >

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context: