similar to: Text to Speech Engine

Where is everyone located? I am in Washington DC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/7c73847d/attachment.htm

didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. Gordon

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

Digital ocean offers ssd on all the virtual machines. Uptime is good. Jai Rangi Www.didforsale.com www.cebodtelecom.com www.cebod.com > On Mar 8, 2015, at 8:11 AM, Jeff LaCoursiere <jeff at jeff.net> wrote: > > > Amazon instances are shared resources. I wouldn't want to count on timing or disk throughput, and you can't just ask them to do "ssd" - its a

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

Hello, Since last few days I have noticed some people complaining that their call gets disconnected while they are in the middle of the conversations. Looking in the log I found these error messages, Dec 10 11:18:56 DEBUG[8833] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/219.206.2.291-089d8768 Dec 10 11:26:41 DEBUG[10410] channel.c: Didn't get a frame from channel:

Hello, I have a issue between asterisk and windows based VoIP system (Client). Vendor SIP Server --> My asterisk --> Client Here is ethereal trace between asterisk and client. 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session description 2 0.042380 192.168.4.23 -> 192.168.3.222

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130509/24815a66/attachment.htm>

HI, I am working as research assistant at South dakota University, in Computing Services Dept. Right now we are working on Suse Linux, We are trying to configure Samba Server on a linux machine. But when we try to open the file/folders in Windows NT machine. Then it ask for user name and Password. Where the global parametrs are set to open to all and all password are set to NO. Now I dont

I was using tagPOS function from openNLP package for parts-of-speach. Now the package is updated and the function is not present. Any suggestions how to do it now ? Thanks for your help. -- Regards, Siddharth Arun, Contact No. - +91 8880065278 [[alternative HTML version deleted]]

Hi Jeff Are you aware of any challenges of hosting it on AWS? It will help me to work out alternate plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance traffic. I see many posts referring to AWS deployment. Please help me to choose AWS server instance. *Thanks & Regards,*

Hello, can anybody interpret what the following means: [root@funke ipp2p-0.8.0]# iptables -t mangle -A MarkList0x666-ipp2p -p tcp -m ipp2p --edk -j MarkSet0x666 iptables: Unknown error 4294967295 ----- I have installed ipp2p-0.8.0 via: make copied ipt_ipp2p.ko to my kernel lib dir copied libipt_ipp2p.so to my iptables lib dir insmod ipt_ipp2p gives the following in dmesg: IPP2P v0.8.0

Dear List, I am wanting to perform some traffic shaping as the subject of this email suggests. What I am wanting to do is this; I would like to have traffic shaping performed on the following protocols: HTTP, RDP, GRE, PPTP, SIP and IAX. Obviously I would like to have highest priority set for voice packets so much so that the general http traffic does not impede on the voice packets. I

Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

// Please reply this email to jai-Oc46A42Di8jtC6uA/Zb986xOck334EZe@public.gmane.org // Hi, My name is Jai, I am a Technical Recruiter with International Business Solutions. Ltd, NJ. My client in NJ is looking for a JD Edwards Technical Consultant. I have included a description of the project below. Please take a few moments to review, and let me know if this is something you might be interested