similar to: outbound calls via google voice not answered by toll free numbers with ivrs

Hello All, My name is Gaurav, and I am tasked with configuring a linux box with NTP. I am having a problem that I hope you can help me with. We have installed CentOS V5 on 1 processor down in the lab. The rest of the processors on the network are AIX boxes versions 3.2.5 and 5.2. I worked through the entire initial NTP configuration; setup the /etc/ntp.conf file to point to the server, etc.

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some

Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110224/2f0d56bc/attachment.htm>

Perhaps someone else has had this issue. I currently have BroadVoice as my SIP provider with two phone numbers. The first number is my primary local number; the second is an 877 toll-free number. The issue is with the IVR abilities. When calling the primary number you can press 200 at the main menu and receive my extension. When calling the 877 number, 200 will result in ?that option is invalid?

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise that this is DATA call, but answering anyway (playing IVR messages, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from 'XXXXXXXX' to '3001' on channel 0/2, span 1

Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that requires you to enter into an ivr system. I already set my dtmf mode in asterisk. --------------

hi, i have a situation where i have a list of options from which a user selects his options if its not in the list he/she selects Others/Not Present and a new textfield should pop up and the data entered there should be considered instead of dropdown. i have done this using javascript/dhtml but i was wondering how this can be done using pure rails i.e using some of the RoR apis?. i would

hi, i have a situation where i have a list of options from which a user selects his options if its not in the list he/she selects Others/Not Present and a new textfield should pop up and the data entered there should be considered instead of dropdown. i have done this using javascript/dhtml but i was wondering how this can be done using pure rails i.e using some of the RoR apis?. i would

Hi all, I'm a beginner user of R. I am stuck at what I thought was a very obvious problem, but surprisingly, I havent found any solution on the forum or online till now. My problem is simple. I have a file which has entries like the following: #ID Value1 List_of_values ID1 0.342 0.01,1.2,0,0.323,0.67 ID2 0.010 0.987,0.056,1.3,1.5,0.4

I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs

I can ping the google.com ip addresses (209.85.231.104) but can't ping the FQDN, obviously DNS is broken. I can also ping the nameserver ip addresses specified in /etc/resolv.conf. -- Gaurav On Sun, Mar 7, 2010 at 10:48 PM, Gaurav Nangla <gaurav.knangla at gmail.com>wrote: > Hi Everyone, > I've setup a Centos 5.4 (x86_64) box in a VM recently; the trouble is I just >

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew

Hi all, I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call on Toll-Free number asterisk accept DTMF digits but dial only first in context. Per instance: When i press 1 it is OK,but when i try to dial extension 700 asterisk dial only first digit(1) and i receive from asterisk "invalid extension 7 in context..".Extensions 700 exists.It seems asterisk dial only

> -----Original Message----- > From: list@ipmotel.net [mailto:list@ipmotel.net] > Sent: Friday, October 01, 2004 12:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] OT: Toll Free > > > Can anyone recommend a good Toll Free provider? I'm looking > for 1 T-1, 2 > Toll Free numbers via VoIP. > NuFone provides us

Hi friends, Is their any possibility to setup our own Toll-Free Number in Asterisk using some PCI or FXO Card? I have one number from my local Telecom called 123XXXXXXXXXXXX and i would like to setup this number in my asterisk if some one called this number from his mobile or land line he should not be charged when the call will come i can route to my SIP or IAX in asterisk internally. In this

FWD (Free World Dialup) allows any SIP call to US toll free numbers via * 18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at FWD so in my dialplan I have something like: exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r) exten => _8.,2,Hangup And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine 99% of the time. I'm wondering if

Hello, since applying the patches related to XSA-36 Xen recognizes a broken IVRS ACPI table and disables I/O virtualisation. I contacted the manufacturer of the mainboard/BIOS and they want to help me by providing a patched BIOS - so far so good. However, they need details about what to fix, which I don''t know either. Could you pls. give me some hints which I can forward to the

Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David