Search found 6 matches

by marcin99
13 Oct 2017 01:38
Forum: 4760
Topic: busy when no free agents
Replies: 6
Views: 712

Re: busy when no free agents

Check the parameter "Dissuasion Busy Tone On DDI" on the pilot attributes
I have enabled this option and receive busy signal only when no agent logged in or when all agents have withdrawal status, but no when all agents are busy (have calls)
by marcin99
12 Oct 2017 09:10
Forum: 4760
Topic: busy when no free agents
Replies: 6
Views: 712

busy when no free agents

Is it possible that configuration?:
Incoming call: Pilot->Queue->PG-> Free Agent
if no free agents: Pilot->Busy Signal or Pilot->Queue->Busy Signal
by marcin99
05 Sep 2016 02:32
Forum: SIP
Topic: Sip trunks on oxe cant register
Replies: 12
Views: 2555

Re: Sip trunks on oxe cant register

if sip is behind NAT you have to:
1. configure destination or static NAT for sip port from provider ip to pbx ip
2. enable sip alg if is disabled
3. configure static nat for rtp from provider to voice card ip for inbound calls
2. configure source nat
by marcin99
05 Sep 2016 02:12
Forum: SIP
Topic: SIP Trunk transfer error
Replies: 8
Views: 2342

Re: SIP Trunk transfer error

@tgn in Asterisk 13 with option: Support Re-invite without SDP is set to TRUE is problem with all incoming calls, I tested it. @ tot3nkopf I used ip address not name. I can use ARS, but then I have to user prefix like for external calls and isn't good solution for my local numbering plan, so it seem...
by marcin99
31 Aug 2016 06:24
Forum: SIP
Topic: SIP Trunk transfer error
Replies: 8
Views: 2342

Re: SIP Trunk transfer error

Hi dryhorse, I checked this options, True/False nothing changed in sip debug. I read many topics and changed trunk type from ABC-F to ISDN All Countries. In this mode transfer works, but I can't add Routing No. or OpenRouting No. so I add Network No. and is ok, but in Network No. I'm able add only o...
by marcin99
31 Aug 2016 03:55
Forum: SIP
Topic: SIP Trunk transfer error
Replies: 8
Views: 2342

SIP Trunk transfer error

hello, I have sip trunk alcatel (OmniPCX Enterprise R11.0.1 k1.520.22.b) - asterisk and everything works well except transfers. When sip call to alcatel and try transfer from sip to other number - work ok. Problem is when I try transfer sip call from alcatel to alcatel, then alcatel send REFER witho...

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