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Nijin Narayanan

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About Nijin Narayanan

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  1. Here is my domain setting page screenshot : https://dl.dropboxusercontent.com/u/7384181/vodia/Domain_Settings.png
  2. Record Call option in missing in 5.2.0 ? We would like to record all incoming and outgoing calls of a particular extension or a trunk.
  3. The system logs are not showing in firefox. When i press the load button, firefox send the GET request to http://pbxaddress/rest/system/log with status 200. But there is no log displayed in screen. I checked in Chrome Browser, its worked as expected.
  4. Current Config: Destination for incoming calls: Send calls to destination in the Request-URI DID Range is 0441003000571 - 0441003000577 Extension & its alias starts from 881/0441003000571- 887/0441003000577 Is the below configuration is correct ? Destination for incoming calls: Match extension after prefix Source for caller-ID: Request-URI Default Account: 70 - AA (Account for Failover Calls) Prefix: 04410030005
  5. We have configured SIP trunk with 5 DID numbers. And incoming calls are routing to its destination based on Request-URI. But we noticed that, For the incoming call from some telecom carriers, our sip trunk provider adding some junk character with its Request-URI. So that we lost those calls. Correct Request-URI is 1234556@ but sometime we got si123456@ & 123456%23@ for some telecom carriers(this was changed few times). We already reported this with them. Is there any way to handle those calls if the PBX is received wrong Request-URI ? to send all those calls to an extension or to IVR ?
  6. Any update on this ? Still getting this Error Message on firefox: "Your browser does not support WebRTC or it is turned off.Therefore WebRTC calls cannot be made.Update or use a browser that supports WebRTC."
  7. No. Others are WAV encoded with A-Law files. So Its a player related issue.
  8. Yes. Its plays fine in VLC. I usually use ubuntu's default player - totem to play all wav files. I can play all other wav files properly except the file received from PBX. It plays like a encrypted audio(something like SRTP enabled calls).
  9. I have enabled the Mailbox message as attachment to email. I'm getting voicemail notification with attachment. But those audio files are encrypted. How do i get the actual voicemail audio ? Can you please check this. Received Audio file: https://dl.dropboxusercontent.com/u/7384181/vodia/msg36.wav
  10. And most of the User login pages are not showing any contents. Like Pages under Advanced -> Features, Account Info, Cell Phone, Notifications etc..
  11. We are getting 400 Bad Request for outgoing calls. Incoming calls are workning. Please help to find the exact issue. Here is the SIP trunk setting: # Trunk 12 in domain testserver.com Name: IP Trunk Test Type: gateway RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: WrtcDestName: WrtcDestNumber: RegKeep: RegUser: Icid: Require: OutboundProxy: Ani: DialExtension: Trusted: false AcceptRedirect: false RfcRtp: false RtcpXr: false Analog: false RtpBegin: RtpEnd: Prack: true SendEmail: UseUuid: false Ring180: false Failover: never HeaderRequestUri: {request-uri} HeaderFrom: {from} HeaderTo: {to} HeaderPai: HeaderPpi: {trunk} HeaderRpi: HeaderPrivacy: HeaderRpiCharging: BlockCidPrefix: Glob: RequestTimeout: Codecs: 0 2 8 18 3 CodecLock: true DtmfMode: Expires: 3600 Fraction: 128 Minimum: 10 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: UseEpid: false CidUpdate: Ignore18xSDP: false UserHdr: Diversion: CoBusy: 500 Line Unavailable CoDest: Colines: DialogPermission:
  12. As i said we have 5 DID number in SIP trunk. So how do i handle incoming call ? Incoming call to each DID should ring respective extension.
  13. We have IP authenticated SIP Strunk with 5 DID Number. How do i configure this SIP trunk on snomone ? & How to handle Outboud & incoming Calls ? I have created SIP trunk with following config: Type: SIP Proxy Proxy Address: SIP trunk IP Address IP Routing List: Any settings needed to work this SIP trunk ? All the Extensions are configued remote location.
  14. We have a hunt group which is configured to ring both ext 222 & 223. Call forwarding in configured on ext 223(in snom phone). If no one pickup the call we need to redirect it to the call forwarding number set on Ext 223. We tried this. But it rings only the ext 222. then the call get disconnected. Any way to archive this call forwarding ? Please advice.
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