similar to: Asterisk SIP URi dialing

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B single port card. When installing the card, the slot on the card doesn't quite line up with the tab in the PCI-E slot. If I loosen the front plate on the card, Ican sort of make it plug in, however, the card won't go in far enough to screw down the plate. I tried the card in the other server and had the same

Dear all, using LinNeighborhood I can view and access all share folders of windows PC. From Network Neighborhood windows Ican see Samba but can't access. Error message is "Network Path was not found". I try using net view \\samba get same error message.

I'm currently porting the public and free parts of Red Hat Documentation to CentOS. Being unable to do anything graphics-related, I need someone to provide the following images: logo.svg 300x140 CentOS Logo image_left.png 124x39 CentOS Logo image_right.png 120x41 CentOS Documentation Logo (to be designed) Thank you! Regards, Andreas -- Solvention Ltd. & Co. KG Egermannstr. 6-8 53359

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

the schedule of my server this configured with -6:00, and this correct one with the normal hour of my country, I made the change but I don't work me [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include

So 4.18.10 has this fix of the bug as observed in that respect, but I think you are seeing a broader example of that issue. NFS4 in this context is not actual NFS, but a type of unix filesystem permission system closely modelled on NT ACLs, deployed in many filesystems Samba uses, as the translation is much more direct. But even without that, it may be we are seeing the same issues, say for

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Dear ALl Can we use Asterisk for only for transcoding?. if yes how many concurent call we can transcode with help of Astetrisk? For this we only need to config SIP.conf or any other file too. Thanks Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100704/f6159f70/attachment.htm

Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in

Hi guys, I setup an IPPBX and some IP Phones to register to it. IPPBX has 6 "register=> xxx@xxx.x.x.x.." sip accounts. When these sip accounts register timeout due to the WAN network problem all the extensions got logon failed. It seems IPPBX register to its own sip proxy server first. If it failed it cannot take the sip clients' request for registration. Anybody has

hi, everyone , could anyone give a exmple of use Ajax.Response? I can not find a sample in prototype homepage --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Spinoffs" group. To post to this group, send email to rubyonrails-spinoffs-/JYPxA39Uh5TLH3MbocFFw@public.gmane.org To unsubscribe from

Everyone, Is there a way to manually assign usb drives to a specified device label. Is there a way to force two usb drives to be labeled as /dev/sdc and /dev/sdd? I decided to build an archive server for the purpose of backing up other fedora/centos desktops at the office. I built a machine and have installed Centos 7.3 on it with all updates current. I also purchased a 3.0 usb sata drive

hi everyone. I am learning RoR and just recently purchase the agile wed dev on rails. i am still at the beginning of the book trying to create the scaffold for the depot_development database. Everytime i run the generate scaffold products admin, i recieve the "before building the scaffold, try to create a table for model products" etc... a few questions: 1. How does ruby know that

to start with i am new to asterisks and i am also a telcom idiot. with that said i have one vonage line i would like to hook up in my soon to be built Asterisk ippbx server. Now with the one Vonage (with call waiting) line can i receive more one call using an auto attendant route the call the approiate extention? thanks mike

*Hi Everybody* ** *I am building a small ippbx network for my office* *I have 6 hard ip phone's and asterisk server but * *now for outging and incoming calls i want to use* *gsm router instead of x100p card ... or pstn* *I want my calls will go out and come through mobile sim card (gsm router).* *My mobile service provider Simultaneously 64 lines conference...* *How many calls can i receive

Hello, I''m getting an URI::InvalidURIError in my controller when I use german characters in the uri. Google accepts them but open-uri not. How to encode them? That''s my controller simplified: def index require ''open-uri'' address = "Bürgerstraße+68+01127+Dresden" api_key = "my api key" json =

Is there any reason for Mongrel to be stripping request.uri as described in this bug report: http://rubyforge.org/tracker/index.php?func=detail&aid=5103&group_id=1306&atid=5145 I was planning on using mongrel to deploy my app but my login code depends on the full request.uri to redirect back after login. Greetings, Pedro.