[asterisk-users] External Recording Server for Asterisk Voicemail

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

[asterisk-users] Asterisk as Text To Speech server

Hi I want to can we use asterisk as TTS server. Which can support mrcpv2 and ssml. Im looking for tts server with above requirement will asterisk 1.8 is useful for me. Any configuration available. Any opensource tts available. Amit-- --

[asterisk-users] Asterisk 1.8 as text to speech server

On Mar 13, 2013 10:16 PM, Amit Salunkhe amitsalunkh...@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

[asterisk-users] Incorrect DTMF detection in Asterisk 1.8

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

[asterisk-users] Alarms Sound files

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

[asterisk-users] Free Alarms sound

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

[asterisk-users] Grandstream GXE2504A codec disable option

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.

[asterisk-users] Asterisk- speech to text(Voicemail to text message)

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

[asterisk-users] Asterisk for transcoding

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-- -- _ -- Bandwidth

[asterisk-users] Hans Rauser

http://shotojukuindia.com/default/index.php -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] How to set call record file name

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-

[asterisk-users] how we can put anybody on hold using Asterisk with analog phone

Hi All we are using Asterisk 1.4.21 users having analog phone connected with Audio codes g/w Mp124. How we can put caller on hold when we receive call on Analog phone (Panasonic). Any dial plan application or feature.conf need to use for this. Audio code g/w having option which we

[asterisk-users] How to access voicemail from deskphone

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

[asterisk-users] Spiral SIP Request problem

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

[asterisk-users] Cisco Phone losse regsitrations with Asterisk

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

[asterisk-users] Asterisk SIP URi dialing

provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit Salunkhe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] How to Barge specific extensions

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

[asterisk-users] How to barge Inbound calls

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

[asterisk-users] InBound call Barging

Hi All I am trying to implement Inbound call Barging using ChanSpy ExtenSpy.Actual requirement is i want to spy or barge Inbound calls received by specfic group or queue. For that i use Set(SPYGROUP) concept but as its inbound calls it playing Inbound channel(whcih is 1st 3 digit of

[asterisk-users] NOKIA E series Phone for SIP-VOIP calling

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

[asterisk-users] Asterisk Meetme its Admin

HI I is there any Application cmd which we can use for Asterisk Meetme confrence for Adminstartor. so, Admin can Manupulate confrence like mute all users kick any user out of the confrence. I tried with Meetme cmd but its not work for me. so is there any idea or config details for this. i

[asterisk-users] Asterisk Meetme its Admin

Hi is there any Application cmd which we can use for Asterisk Meetme confrence for Adminstartor. so, Admin can Manupulate confrence like mute all users kick any user out of the confrence. I tried with Meetme cmd but its not work for me. so is there any idea or config details for

[asterisk-users] Asterisk Meetme MeetMeAdmin cmd info-use

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

[asterisk-users] Asterisk call waiting with SIP

Hi any body have idea about asterisk call waiting with SIP if we use asterisk1.4.5 hard phone with extensions. i need dial plan logic for this which capable to activate deactivate such feature. with ATA IP phone it is possible but with normal hardphone SIP in asterisk is it possible