[Where is Record Call Feature in 5.2.0]

Here is my domain setting page screenshot : https://dl.dropboxusercontent.com/u/7384181/vodia/Domain_Settings.png

[Where is Record Call Feature in 5.2.0]

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.

[System logs are not showing in Firefox - Vodia 5.2.0]

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.

[Incoming Call Failover]

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

[Incoming Call Failover]

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@1.2.3.3 but sometime we got si123456@1.2.3.4 & 123456%23@1.2.3.4 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 ?

[Web interface strange pages]

Its fixed in 5.2.0

[webrtc / firefox]

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."

[Mailbox message as attachment to email: Getting encrypted Audio file]

No. Others are WAV encoded with A-Law files. So Its a player related issue.

[Mailbox message as attachment to email: Getting encrypted Audio file]

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).

[Mailbox message as attachment to email: Getting encrypted Audio file]

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

[Web interface strange pages]

And most of the User login pages are not showing any contents. Like Pages under Advanced -> Features, Account Info, Cell Phone, Notifications etc..

 

[IP Authenticated SIP Trunk: 400 Bad Request for Outgooing Calls]

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: 242.59.17.5:5060

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:

[How to configure IP Authenticated SIP trunk]

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.

[How to configure IP Authenticated SIP trunk]

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: 172.31.0.0/255.255.0.0/172.31.13.89 0.0.0.0/0.0.0.0/Public_Static_IP

 

Any settings needed to work this SIP trunk ?

 

All the Extensions are configued remote location.

[Call Redirect to Cell Phone when no one pickup the call]

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.

[Enable RTP Phone to Phone Communication]

Hi,

I have noticed that for the internal call (extension to extension), RTP communication is via PBX.Both the snom phones are in same network.In normal case RTP communication is end-to-end communication.

 

How can we disable this RTP via PBX ?

 

Please advise.

 

Thanks,

NN

[Hide Corporate Extensions from Directory Search]

Is that possible to hide the Corporate level Extension from snomone's directory search ?

 

Eg: We need to hide the CEO's Extension fron the directory list but CEO need to see all other extensions too.

[Extension's Webinterface Username & Password]

I was unable to access the Web Interface of snom IP phone which is configured via PnP. What will be the default web interface username and Password ?

 

I tried the all the password i have given during the creation of extension. but it fails.

 

Please help.

[IVR and Voicemail announcement are not hearing for first few seconds]

We have setup the snomONE with Sangoma A101D Card. And Incoming call to the ISDN trunk is forward to Auto Attendant. When I dialed the number, the call was connected to IVR. But i cant able to hear IVR announcement for firt 25sec.

If I forward the calls to and extension, in this case both party can hear well. But if the incoming call was reached to Voice mail box then same issue occur.

 

This will not happen when an extension dial IVR. It works as expected.

 

Is there any setting need to change ?

[Dynamic IP with snomONE behind NAT]

Why snomONE support DNS name is IP Routing list ?

 

We have a dual wan router and it set to auto-switch to a wan if another was goes down. SO every time i need to change the IP address in snomone.

 

Is that possible to setup an Crontab in snomONE soho to update the IP routing list ?