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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 serv...

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

...ybody 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 Salunkhe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081223/e1982f09/attachment.htm

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

...oip-info.org/wiki/index.php?page=Asterisk+variables * http://www.voip-info.org/wiki-Asterisk+cmd+dial I think, the lines next to t option are incomplete. Variable GOTO_ON_BLINDXFER was obsoleted at some point (probably 1.4), i'm not sure that it will work in 1.4 Regards, Atis On 9/13/07, amit salunkhe <amitsalunkhe21 at gmail.com> wrote: > Hi sir > i read your response mail on asterisk user list i need help with > that dial plan which help if we transfer the call to any extension & it not > anser with 26 sec, then that transfer call come back to us . > so...

...cemail(u113) exten => 2,4,hangup exten => 3,1,playback(gerencia) exten => 3,2,Dial(SIP/112,12,Tt) exten => 3,3,voicemail(u112) exten => 3,4,hangup [cyber] include => internacionales include => internas include => locales include => parkedcalls greetingss From: amit salunkhe <amitsalunkhe21 at gmail.com> HI as per you in context ur inclusion is correct . but if it is time missmatching as per your ur office shedule then you have to check which time period show by ur server according to that you have to set. As asterisk server normally follow GMt time then u...

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

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 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

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

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

...; 999,103,Wait(1) exten => 999,104,Goto(1) [softoption-gradwell] exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,) exten => _0[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:1},,) exten => _[1-9]XXXXX,1,Dial(IAX2/Gradwell/441353${EXTEN},,) ________________________________ From: amit salunkhe [mailto:amitsalunkhe21 at gmail.com] Sent: 14 February 2008 07:44 To: Phil Knighton Subject: UK -999 dialing issue HI Phil Can u send me ur out call context config. Also tell me what ur using with Asterisk to make out call SIp-Voip or Pstn line with Fxo card? also check w...

Hi All I need to set my Asterisk conference such way that , during confernce Admin Can kick 1 or all user , Same for mute fuction.As well as Admin can increase or decrease conf & user volume. for that i used MeetMeAdmin like this exten => 600,1,MeetMeAdmin(1111,ekKLmMNS,7010) where 1111 is conf number & 7010 is Admin user

Hi All Can anybody help me for dial plan which can barge inbound call groupwise. Because when i am trying to barge inbound calls which is coming on my DID number i can hear 1st 3 digit of my Inbound provider IP address instaed of extension which pick that calls. I tried Chanspy as well as Extenspy. But result is same. So Plz Help me. Thanks Amit -------------- next part --------------

Hi We are using Asterisk + java based Pd Dialer. Cisco 7040 IP Phone we are using as extensions or Agent phones. currntly we set NAT keep alive time less as possible & registartion time= 25 sec instaed of 3600. But following issues we are facing & im not sure whther its due to internal netwokr issue where that Phoen are placed or Asterisk or java application issue. Porblem is when

Hi All we are using Asterisk 1.6.0.9 version.try to use Minivm for voicemail, but having following problem. 1.How any extension let's say 7001 can access his voicemail box from his deskphone, any config or dial plan example is there. What kind of config require in extension.conf,minivm.conf & sip.conf 2. same way for MWI on respective IP Phone what we require to config in

Hi All For recording inbound call we are using following line in dial plan.But we wish to set file name which describe who attend the call or lets say extension of the call attendant. Current line in dial plan to set file name is like this- *Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}) *which results after completion is "time-calledid

Dear All Among all the readers anybody have ever work on Granstream device GXE2504A which act as ippbx and having GUI to configure and maintain. We are facing one problem with this device, thsi device reply or adding codec like ilbc,G.721 which is not supported by our Asterisk server or our SBC. We want to disable this codecs, but form available GUI we not able to see any option to disble it.

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110509/cc00bd60/attachment.htm>

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110516/c4d3be03/attachment.htm>