similar to: External Recording Server for Asterisk Voicemail

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

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

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 i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

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

Dear All Can you let me know is this possible to if we are using Asterisk version 1.4 or 1.6 for incoming voicemail we can send as email in text formta. Means voice mesage converted into text message & send it to resp. email ids. is this possible. If yes. we can do the same with help of Asterisk or we require expertnal application need to isntall/integrate to work for speech to test. Please

Hi All, I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global default settings. but when user sending DTMf event with SIP info method my asterisk accepting that DTMF. If default or global setting is rfc2833 then how come asterisk accepting SIP info dtmf event? what to check please guide Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

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

Hi all .. Maybe I am just missing something, but when I press '0' and then '3' to record my name, it gives me an 'after the tone .. '. Then I say my name and press #. It says: your message has been saved. So HOW do I listen to my recording, make sure it sounds OK, and then CONFIRM that is what I want the mailsystem to use OR CANCEL out and leave things the way they are?

Hi, I'd like to trigger call recording during call. Do I have any keys that can be pressed during call ? I've tried this, but doesn't start anything ( I guess that "a" is active only during voicemail ?): exten => a,1,DBget(temp=Record/${TIMESTAMP}_${UNIQUEID}_${CALLERID}) ; Already recording ? if not goto 102 exten =>

Dear friends, please excuse me if my question will be trivial. I've installed and started Asterisk (stable 1.0.7, but with CVS HEAD I experienced just the same problem), and changed a bit sip.conf: [general] ; ... dtmfmode = inband disallow = all allow = ulaw allow = alaw allow = gsm run kphone, and call the 1235 extension. According to sample extensions.conf, Asterisk would

Hello All, Our old Lucent Argent system had a feature whereby when you initiate recording during a call, it would afterwards send the recording as a voicemail message to the user who initiated the recording. We use the automon *1 recording function in asterisk, which allows users to record a call if necessary on the fly. Unfortunately there doesn't appear to be an easy way for the user to

Hello, I am sending email notification when new voicemail is received, with the voicemail message attached. Therefore, once this email is sent, I don't want to keep the original voicemail message on the asterisk server, as the user does not need to call in to listen to the message. Once the email has been sent, the message should be deleted and counter reset. At the moment, the the

Hi. Back before Christmas Arnaud sent me a patch for the snmp-ups driver for my HP R12000/3 UPS. Unfortunately shortly after I had to take quite a lot of sick leave so wasn't able to progress it, but I'm working on it again now. I'm at the point where I can start the driver with upsdrvctl and start the daemon with upsd. When I check the UPS status though it shows as OL DISCHRG OB,

Dear asterisk users, I am presently playing with an asterisk@home. I am trying to find the best codec solution for my voicemail records. I want to use ARI (Asterisk Recording Interface) to read the messages. I first used the default wav encoding that was not appropriate because my navigator does not handle wav mime types correctly and I have difficulties playing wav files in my basic linux

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: