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Some tips or a working configuration for an Audiocodes Mediant ESBR running the latest v7.20 firmware would be good. I found pre-v7 firmware quite easy to understand but the new firmware interface (where it looks better) does not give me the same results for setup - I am running v7.20...

Hi Drew, Thanks for responding and yes I achieved this with (I have x'd out my DID for obvious reasons): [from-trunk] exten=>xxxxxxxxxxx,1,Answer exten=>xxxxxxxxxxx,n,Goto(digitalreceptionist,s,1) exten=>xxxxxxxxxxx,n,Hangup Digital receptionist being: [digitalreceptionist] exten =>...

Hi all, Asterisk CLI: This one should be fairly simple but when I dial a number and it fails (internal mostly) I want to replace the audio file that is played generically by the system. This file sounds like an engaged tone. In Asterisk I get "rejected because extension not found" in the debug...

Hi, Yes, I have retested, although both files with the "file" command present the same credentials if you don't export as you have said it does not work - its working now however - thanks for your help!

Thanks for coming back with your response, I have tried out your suggestions but with no success, it has made a slight change to the error messages I now see. Although I get similar results I now also see this starting RIFF result: format_wav.c:164 check_header: Does not begin with RIFF...

Hi, Thanks for coming back to me. The file has been uploaded in a few ways - both by this online file formatting tool: http://my.digium.com/en/products/ivr/audio-converter/ And by the guidelines outlined here: http://wiki.kolmisoft.com/index.php/Convert_WAV_file_to_Asterisk_playable_format I...

Hi, Could someone please give me assistance with this issue. I am running Asterisk 13 and have setup an IVR, the extensions.conf call flow seems to work fine up until the point of playing my pre-recorded audio file. The below error(s) are seen in the Asterisk console no matter what settings I...

Hi, Can someone tell me the context to routing inbound calls from a trunk/DDI to a call queue (assign a Queue or IVR a DDI to call basically). My call queue is setup in queues.conf and works, it has an internal number of 6400 (which works if I dial it) however I have tried mapping a DDI in...

Solved with: exten => _X.,1,Set(CALLERID(num)=SIP-ID) exten => _X.,n,Dial(SIP/${EXTEN}@sipgate,30,trg) exten => _X.,n,Hangup

I have setup an Asterisk version 13 system to a 3CX system (similar to the below guide) - it registers fine: http://www.3cx.com/docs/bridging-asterisk-pbx/ The issue I have now is the routing of calls of which I am sure the dialplan is the answer. If I call from 3CX to the bridge (Asterisk ext...

Hi Ucxguy, See my previous post - you will see I put a note in to say ignore this, the trunk ID and password I have changed for that very reason - don't worry, I will try your suggestion however I think I may of already tried this. Tell me - why is it saying not from internal when my settings...

Hi, I am relatively new to Asterisk and currently getting to know the CLI. I have a basic setup of which internal calls work without issues and I have a Sip Trunk registered with SIP Gate. If I try to make an outbound call however I get an error that: Call from '6001' (10.5.10.10:5060) to...

I have tried all of the suggestions - thanks for them they have been useful with my understanding. Still experiencing the same issue, one question regarding the logging - is there anyway to identify which Shortcode is being used for the call ? I can see it is selecting the correct trunk however...

Thanks for the response, I understand this better now - so instead of "9" in this case, how would I have it say if I dial anything starting with a "0" with an 11 digit number (for example) send it out via my trunk ?

Please elaborate more on the shortcode, it seems to work, the number dials and I see it come up in the trunk when I run a trace. - that would tell me that the PBX is dealing with the call and sending it on. I am not new to PBX's but am to Avaya IPO so please let me know what the issue is with...

Hi, Yes, I did think of this myself and have tested in international format also, I get exactly the same result. Thanks.

This is with just SIP tx, rx + verbose mode: ********** SysMonitor v9.1.7.0 build 163 [connected to 172.30.0.250 (IPOfficeDemo (Server Edition(P)) (Select))] ********** INVITE sip:01635268937@sipconnect.sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP...

Thanks for responding, see below for the full trace: ********** Warning: Logging to Screen Started ********** 10893770mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,201: Digit Key Pressed 0 10893770mS CMCallEvt: 0000000000000000 0.1043.0 -1 BaseEP: NEW CMEndpoint f4227b38 TOTAL...

I have an IPO version 9.1 setup with some 9620 phones and a SIP Trunk from SipGate. Phones are registered and inbound calls work without issues. It is my outbound calls that fail with a "call rejected" being displayed on the phone. It is matched with a shortcode of: 0N/Dial/9 The monitor...