similar to: Asterisk + voxbone ==> Failed to authenticate user

Hi List, I recently signed up with Voxbone to get some International DIDs. I was just about to purchase a DID this morning... but when I went to get it.... voxbone wanted the end user's address information. So I started to put it in... unfortunately... the end-user is in the U.S....but the only options are for a few select cities in GERMANY! I don't understand. Is there some

Hello, In about one hour we should be chatting with Tim Behrins of Voxbone about their initiative, iNum. I say "should" because he's the scheduled guest, but I haven't heard from him today :) Next week, we'll be "Hacking VoIP" Feel free to top post your answers, it seems to stimulate conversation. /r http://VoipUsersConference.org for the usual data or jump on

Hi, We have been using Voxbone for some time for origination, and they now offer E911 services.? We are trying to set this up and having trouble meeting their authentication requirements. I setup a peer as I normally would, with user/pass as they supplied ("lacoursj", "pass"), but my calls are rejected.? Their support is asking that I follow this auth mechanism: 1st step

I *am* doing that, as I assumed it would be required just for the 911 mapping we have provided, but that doesn't change the SIP header. Cheers, j On 05/08/2018 02:41 PM, Khalil Khamlichi wrote: > try setting the callerid with > > same => n,Set(CALLERID(all)=17864089672 <17864089672>) > > ofcourse for each customer you will need to provide his own did. > > >

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

All, I'm in Dallas, TX, US and am looking for inbound-only DID service with 10+ channels on a SIP trunk. Is anyone on this list doing something similar and have any recommendations for a provider? Of course I'll be routing either SIP/IAX to an asterisk server that will be hosted in a Dallas colocation facility. I found VoxBone.com but they only want to deal with customers that can

Dear, Currently, I'm looking for a UPS that can support NUT. I found an APC Smart-UPS 5000VA 230V SRT, but I'm not 100% that is supported. *APC Smart-UPS 5000VA 230V SRT* http://www.apc.com/shop/nl/nl/products/APC-Smart-UPS-SRT-5000VA-230V/P-SRT5KXLI I searched in http://networkupstools.org/stable-hcl.html and I found that you support "*APC Smart*" I would just to confirm

Hello, I want to place an Asterisk-server A in front of 2 other Asterisk-servers (B1 & B2). This first Asterisk-server A needs to send incoming calls to one of the 2 available Asterisk-servers (B1 or B2) behind it. So I want the first Asterisk-server A to accept the call, and based upon some checks in the dialplan send the call through to one of the other Asterisk-servers (B1 or B2)

Hello: Some time ago, in this list, I asked for a DID provider in Spain. Somebody answers me detailing his company DID services. I has an email problem and lost his response and his mail address. Please, if you (the DID provider) are reading this, let me know again your mail address because we are going to contract your DID service. Regards

I know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for

I am looking for a retail DID provider which should provide unlimited incoming calls something around 4-5 bucks . Nufone seemed like a good choice at $5 but they are down again :( . Any suggestions please . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070802/087c4117/attachment.htm

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Post the following please. Thanks! Voxura, an internet video sharing startup, is looking for a skilled and motivated Web User Interface Developer to work with us in building our video sharing products and features. You will work with a strong, creative team building features used by people every day. Our team is looking for a Web User Interface Developer with in-depth knowledge of web

Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've done that to improve security and to be able to assign all calls coming in

Hi! OS: CentOS Linux release 7.1.1503 (Core) Dovecot: 2.2.18 Run user listing - doveadm user '*', single user - work. auth[20540]: segfault at 8 ip 00007fd554340c6b sp 00007ffc17c886e0 error 4 in libauthdb_ldap.so[7fd55433b000+a000] dovecot: auth: Fatal: master: service(auth): child 20540 killed with signal 11 (core dumped) dovecot: auth: Error: net_connect_unix(auth-worker) failed:

I heard from a Voxbone executive that they are opening a new NOC near Los Angeles and probably hiring a few voip support engineers. If anyone is interested feel free to contact them through their website Cheers, Robert ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites

Thanks to Digium, the company, and to all of the fine people from Digium who participate in the weekly VoIP Users Conference conference! We will be live on Friday January 1, 2010 and there is also a "reel" of recorded greetings from people around the world wishing the VoIP Community a Happy New Year. You can hear this anytime during the year by downloading it from the site starting next

Hi, Tomorrow, our discussion is around iNum with lots of interesting people chiming in, including the Voxbone people who manage the space. If you ever wondered about iNum and why you might care about it, how it works, who offers it and who actually uses it, here's a chance to find out more. Join us live at 12 Noon EST or find the local time for you at http://vuc.me/next Call: sip:200901 at