vicidial.org

I have a situation where I would like to have all of my closers do a transfer to a secondary closer campaign (offsite) it will reach this closer campaign via a DID on a different system. Is this viable?

Sure, but not if you want to send customer information along with the call. There needs to be something that links the two if you want that.

If you use IAX2 trunks you can do this, but not with normal PSTN DIDs.

Heres what Im running into...

Scenario 1 (Blind Transfer)
1. Call comes in via an IAX provider..
2. I goto Transfer-Conf and select blind transfer
3. the caller gets hung up on and a call is never initiated to the person Im attempting to transfer the call to.

Scenario 2 (Dial with Customer)
1. Call comes in via an IAX provider..
2. I goto Transfer-Conf and select dial with customer
3. Call goes out of a different SIP provider to the 3rd party
4. 3rd party answers call
5. Agent selects "Leave 3-way call"
6. 3rd party gets hung up on immediately.
7. customer never gets hung up and stays connected to the agent but the agent goes through disposition like it does not have a live call even though it does.

am I missing something?

First scenario, please post some Asterisk CLI output when you do this.

Second scenario, do you have available "conferences" table entries and corresponding meetme rooms available(not vicidial_conference, but conferences)?

I would prefer to not even deal with the second option. I believe the issue with option one is that its trying to issue and "X" before dialing the number since "within my campaign settings, I have X as a pre-value for dialing so that it does not dial a 9 etc".

Code: Select all

Feb 22 14:22:51 WARNING[32013]: pbx.c:2357 __ast_pbx_run: Channel 'SIP/telespan-0087f370' sent into invalid extension 'X1602xxxyyyy' in context 'default', but no invalid handler


What is the work around to this?

If this is for blind transfer, you just need to check the OVERRIDE checkbox to only dial the exact extension you put into the number-to-dial field.

Yea got that working yesterday however I have no audio between customer and 3rd party..

Agent gets removed from call
Customer heads ringing in handset
3rd party gets rung and answers

no audio

Any Asterisk CLI output on this?

yea sorry forgot to post it..

initial call

Code: Select all

    -- Executing AGI("Local/16026638055@default-cf9e,2", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/16026638055@default-cf9e,2", "SIP/telespan/16026638055||o") in new stack
    -- Called telespan/16026638055
    -- SIP/telespan-0089b660 is making progress passing it to Local/16026638055@default-cf9e,2
    -- SIP/telespan-0089b660 answered Local/16026638055@default-cf9e,2
       > Channel Local/16026638055@default-cf9e,1 was answered.
    -- Executing AGI("Local/16026638055@default-cf9e,1", "agi://127.0.0.1:4577/call_log") in new stack
  == Spawn extension (default, 16026638055, 2) exited non-zero on 'Local/16026638055@default-cf9e,2'
    -- Executing DeadAGI("Local/16026638055@default-cf9e,2", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing AGI("SIP/telespan-0089b660", "agi-VDAD_LB_transfer.agi|8365") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_LB_transfer.agi
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing DeadAGI("Local/16026638055@default-cf9e,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----0") in new stack
    -- AGI Script agi-VDAD_LB_transfer.agi completed, returning 0
    -- Executing MeetMe("SIP/telespan-0089b660", "8600153") in new stack
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----0 completed, returning 0
  == Parsing '/etc/asterisk/manager.conf': Found


call transfered

Code: Select all

    -- Executing Dial("SIP/telespan-0089b660", "SIP/telespan/16024540471||o") in new stack
    -- Called telespan/16024540471
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing DeadAGI("SIP/telespan-008176e0", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing DeadAGI("SIP/telespan-008176e0", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----6") in new stack
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----6 completed, returning 0
    -- SIP/telespan-0085d3b0 is making progress passing it to SIP/telespan-0089b660
    -- SIP/telespan-0085d3b0 answered SIP/telespan-0089b660
    -- Attempting native bridge of SIP/telespan-0089b660 and SIP/telespan-0085d3b0


I don't see any errors on the transfer.

One problem might be with the native SIP transfer, you might want to try a non-native transfer by putting a 't' in the dial flag:

||o changed to ||to

thats the ticket.. thanks for the help!

Another thing I've noticed specifically in CLOSER campaigns is that the D1 D2 options do not work even with the number specified in the campaign settings. Is there something specific to closer campaigns? I didnt notice anything in the code that would stop this.

The variable is being passed on the page

var CalL_XC_a_Dtmf = '1XXXYYYZZZZ';
var CalL_XC_a_NuMber = '1XXXYYYZZZZ';

What exactly happens when you click on the D1 or D2 links?

Are there any Javascript console errors?

absolutely nothing happens there are no errors not warnings..

I have the number populated inside the campaign for both

Transfer-Conf-number-1/2
Transfer-Conf-DTMF-1/2

Is this only on outbound calls when in a CLOSER campaign?

I just tested inbound calls and they take on the presets of the in-group of the inbound call.

No actually all of my outbound campaigns take it just fine but my inbound closer campaign doesnt take it..

That's very odd, I just tested that with a couple inbound groups and their presets came over with the call just fine, what version of astguiclient are you using?

Excuse me for this insolent question, but:
What is the point of filling the same number in DTMF and NUMBER?

enjay wrote:Another thing I've noticed specifically in CLOSER campaigns is that the D1 D2 options do not work even with the number specified in the campaign settings. Is there something specific to closer campaigns? I didnt notice anything in the code that would stop this.

The variable is being passed on the page

var CalL_XC_a_Dtmf = '1XXXYYYZZZZ';
var CalL_XC_a_NuMber = '1XXXYYYZZZZ';