vicidial.org

by **kolucoms6** » Mon Jan 05, 2009 12:12 pm

I have been following Manager Manual to create an Inbnd Camapign to recieve Inbnd Calls.

When I logged in with cc101, as per manual,

Code: Select all: Because you logged into a CLOSER-enabled campaign, you should now see a green screen with the listing of all in-groups. Click on the SALESLINE group in the left side column and then click SUBMIT at the bottom of the screen. There is also a BLENDED check-box on this green screen. You would select that if you want to do blended calling where outbound calls are placed from that campaign while agents are also taking inbound calls.

I see a green screen which goes away in 2 secs but that green screen shows a drop down box with "select a group to send your calls to". When I tried to click on it, it shows nothing.

Then I see a window with

BLENDED CALLING(outbound activated) .I check that.

Now when I try to call my own number , call doesnt comes in.

Kindly advice.

Also, my US DID number is 20620368XX.So, I have made an Entry in Extension.conf file as

exten => 20620368XX,1,Ringing ; call ringing
exten => 20620368XX,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 20620368XX,3,Answer ; Answer the line
exten => 20620368XX,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-----LB-----SALESLINE-----20620368XX-----Closer-----park----------999-----1-----TESTCAMP)
exten => 20620368XX,5,Hangup

at the end of the file. Is that Ok ?

by **mflorell** » Mon Jan 05, 2009 5:23 pm

Is the in-group you created in admin.php selected in the checkbox in the campaign detail screen at the bottom(allowed in-groups)?

by **kolucoms6** » Tue Jan 06, 2009 3:20 am

Yes.

Code: Select all: Allowed Inbound Groups: SALESLINE - Primary Sales Line Default Transfer Group: SALESLINE - Primary Sales Line ---NONE--- Allowed Transfer Groups: SALESLINE - Primary Sales Line

by **mflorell** » Tue Jan 06, 2009 8:50 am

So you don't see SALESLINE anywhere on the screen with BLENDED CALLING checkbox on it?

by **kolucoms6** » Tue Jan 06, 2009 9:50 am

No.

Here are Screen shots :

by **mflorell** » Tue Jan 06, 2009 7:31 pm

Wrong screen, do you have SALESLINE checked in the

CAMPAIGN DETAIL screen?

by **kolucoms6** » Wed Jan 07, 2009 12:44 am

by **mflorell** » Wed Jan 07, 2009 2:03 am

Well, that should be good then, if you are still having problems please post some output from the agiout logfile when the calls go through the ALL_inbound AGI script.

by **kolucoms6** » Wed Jan 07, 2009 6:37 am

Entries in Extension.conf file is Ok ?

by **kolucoms6** » Wed Jan 07, 2009 10:31 am

2009-01-06 18:10:57|agi-VDADtransfer.agi|Perl Environment Dump:
2009-01-06 18:10:57|agi-VDADtransfer.agi|0|8365
2009-01-06 18:10:57|agi-VDADtransfer.agi|callerID changed: V0106181027000070719
2009-01-06 18:10:57|agi-VDADtransfer.agi|AGI Environment Dump:
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- accountcode =
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- callerid = unknown
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- calleridname = V0106181027000070719
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- callingani2 = 0
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- callingpres = 0
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- callingtns = 0
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- callington = 0
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- channel = SIP/sip-08b05430
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- context = default
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- dnid = unknown
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- enhanced = 0.0
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- extension = 8365
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- language = en
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- priority = 4
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- rdnis = unknown
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- request = agi-VDADtransfer.agi
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- type = SIP
2009-01-06 18:10:57|agi-VDADtransfer.agi| -- uniqueid = 1231283427.1273
2009-01-06 18:10:57|agi-VDADtransfer.agi|AGI Variables: |1231283427.1273|SIP/sip-08b05430|8365|SIP|V0106181027000070719|V0106181027000070719|4|
2009-01-06 18:10:57|agi-VDADtransfer.agi|+++++ VDAD START : |70719|2009-01-06 18:10:57|1.2.27|4|
2009-01-06 18:10:57|agi-VDADtransfer.agi||SELECT count(*) FROM vicidial_live_agents where callerid='V0106181027000070719';|
2009-01-06 18:10:57|agi-VDADtransfer.agi||SELECT count(*) FROM vicidial_auto_calls where callerid='V0106181027000070719' and status IN('LIVE','XFER');|
2009-01-06 18:10:57|agi-VDADtransfer.agi|-- VDAD : |0E0|update of vac table: V0106181027000070719
|UPDATE vicidial_auto_calls set uniqueid='1231283427.1273', channel='SIP/sip-08b05430',status='LIVE',stage='LIVE-0' where callerid='V0106181027000070719' order by call_time desc limit 1;|
2009-01-06 18:10:57|agi-VDADtransfer.agi||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='192.168.0.2' and campaign_id = '' and call_time < "" and lead_id != '';|
2009-01-06 18:10:57|agi-VDADtransfer.agi|WWWWWWWW VDAD XFER WAIT: |0||SIP/sip-08b05430|V0106181027000070719|1231283427.1273|

calls coming into this number to go to VICIDIAL

Was also wondering , I dont have to configure Inbound SIP Accnt in the Dialer ?I am using Pulver and I have configured Eyebeam to recieve Inbound call .

[/code]

by **mflorell** » Wed Jan 07, 2009 6:56 pm

That agiout logfile output was for an outbound call.

by **kolucoms6** » Thu Jan 08, 2009 11:30 am

This is what I have done :

1) Create In-Groups as SALESLINE
2) Created Campaign as TEST_IN
3) Enter suggested lines in Extension.conf.
4) Logged in and selected campaign as TEST_IN
5) Selected Blended Call Mode.
6) From my xlite, I tried to dial my own Inbound number i.e 2062036895.

My Outgoing and Incoming , both are different SIP providers.

How will Call understand that it needs to hit Vicidial ?[/b]

Just to try, I made an entry in sip.conf file :

[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
type=peer
context=from-fwd
host=fwd.pulver.com

Is this correct ?

by **kolucoms6** » Thu Jan 08, 2009 11:59 am

Also, I see No entry in agiout log file for agi-VDAD_ALL_inbound.agi.

by **kolucoms6** » Fri Jan 09, 2009 4:31 am

Matt, any further guidance ?

by **arvindsandilya24** » Fri Jan 09, 2009 5:31 am

Check with this Dial plan . please change ingroupid in dialplan.

exten => 20620368XX,1,Ringing ; call ringing
exten => 20620368XX,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 20620368XX,3,Answer ; Answer the line
exten => 20620368XX,4,AGI(agi-VDAD_ALL_inbound.agi,CID-----SO-----ingroupid)
exten => 20620368XX,5,Hangup

Thanks & regards
Arvind

by **kolucoms6** » Fri Jan 09, 2009 1:29 pm

Sorry, but not working.

Any further guidance ?

by **kolucoms6** » Fri Jan 09, 2009 1:50 pm

ingroupid is hardcoded as ingroupid or its some value ?

by **kolucoms6** » Sat Jan 10, 2009 7:19 am

What am I missing that I cannot solve it ?

by **kolucoms6** » Sat Jan 10, 2009 2:27 pm

Sorry for bumping... But I am stucked somehere but dont know where

by **williamconley** » Sat Jan 10, 2009 10:00 pm

you need to follow standard "inbound SIP call" procedures for asterisk so that the call will arrive at the context [default] if that is the context you are using.

after that, here is a sample of the setup for the inbound in extensions.conf in [default]:

exten => 4075060000,1,Ringing()
exten => 4075060000,2,Wait(1)
exten => 4075060000,3,Answer()
exten => 4075060000,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----4075060000-----Closer---------------999-----1)
exten => 4075060000,5,Hangup()

After that you create "SALESLINE" in "ingroups" to link this dialplan entry to the programming in Vicidial.

by **kolucoms6** » Sat Jan 10, 2009 10:09 pm

But I need any entry in SIP.conf file ?

by **williamconley** » Sat Jan 10, 2009 10:27 pm

If SIP is how the call is coming in, then yes. The carrier must be connected to asterisk and the call must arrive at the [default] context... so you would use the sip.conf to specify that is where the call goes.

in sip.conf: (this is just a sample, your settings will be different)

[provider]
context=default
type=friend
qualify=1000
host=xxx.xxx.com
dtmfmode=inband
allow=all
nat=1

by **kolucoms6** » Sat Jan 10, 2009 10:31 pm

When you refer to DeFAULT context , what are you referring to?

My SIP entry is :

[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
type=peer
context=from-fwd
host=fwd.pulver.com

I hope its correct.

I am asking this as there is NO MENTION of SIP.conf file in Manager Manual for Inbound Configuration.

Also, agi-VDADcloser_inboundCIDlookup.agi (Your suggestion) OR agi-VDAD_ALL_inbound.agi (Manual suggest this to use it), which one should I use ?

by **williamconley** » Sat Jan 10, 2009 10:34 pm

I think the fact that it is an inbound call, the writer may have assumed that you'd be bringing the call in to asterisk as part of the process.

you will still need to specify the [context] (in my sample it was context=default

this will tell the system where to send the call, and when it arrives there it will look for the DID if you have set it up properly. send it to wherever the rest of your vicidial dialplan is. not to something internal within your asterisk plan.

by **kolucoms6** » Sat Jan 10, 2009 10:38 pm

So, I presume that Manager Manual basically doesn't get into much of Asterisk. Sorry, I was NOT aware of it.

I need to get one Asterisk Manual :lol:

Also, I don't see SALESLINE anywhere on the screen with BLENDED CALLING check box on it. Any mistake Am I making ? Screen shot I have attached ?

Can you suggest me a good site for Asterisk tutorial

by **williamconley** » Sat Jan 10, 2009 11:21 pm

Strangely enough, I DO see "SALESLINE" in your screenshot.

It's in the definition of the Campaign: In allowed Transfer Groups and Default Transfer Group.

You also need to be sure that the user is allowed access to the SALESLINE inbound group as well. This would be in the definition of the user.

by **okli** » Sat Jan 10, 2009 11:33 pm

kolucoms6 wrote:Can you suggest me a good site for Asterisk tutorial

http://www.voip-info.org/

Just about anything you need to know about asterisk.

I use it most of the time along with this book:
http://www.amazon.com/Asterisk-Telephon ... 355&sr=8-1

and google of course.

by **kolucoms6** » Sat Jan 10, 2009 11:33 pm

I am referring to the green screen shot above where I shld have an option to select a campaign as per managers manual..

by **mflorell** » Sun Jan 11, 2009 5:12 am

You don't see it because "Manager has selected groups for you" You specifically prevented the agent from selecting groups.

by **kolucoms6** » Sun Jan 11, 2009 9:50 am

Which group are we talking about here ?

by **kolucoms6** » Mon Jan 12, 2009 12:27 am

You also need to be sure that the user is allowed access to the SALESLINE inbound group as well. This would be in the definition of the user.

I have selected that but still I dont see any group to select from.

by **mflorell** » Mon Jan 12, 2009 2:00 am

Your problem is that you changed "Agent Choose In-groups" to 0

by **kolucoms6** » Mon Jan 12, 2009 2:08 am

Cool.

Now I see that group.

[OFFTOPIC]

But now my concern is I have an account with FWD and I have configured my SIP.conf with

[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com

But when I am trying to dial out my own DID , I dont see any call landing in asterisk.

In extension.conf file I have

exten => 2062036895 ,1,Ringing()
exten => 2062036895 ,2,Wait(1)
exten => 2062036895 ,3,Answer()
exten => 2062036895 ,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----2062036895-----Closer---------------999-----1)
exten => 2062036895 ,5,Hangup()

2062036895 is with IPKall.

What could be the wrong. ?

by **kolucoms6** » Wed Jan 14, 2009 12:57 am

Finally, my call is landing on my Asterisk server . I changed the SIP Proxy in IPKall site as my Static IP.

But when I logged in as cc101 , I dont see any call coming in.

CLI :

login as: root
root@192.168.0.2's password:
Last login: Tue Jan 13 19:24:19 2009 from 192.168.0.21

Welcome to VICIDIALNOW!!!
-------------------------------------------------

For access to the VICIDIAL admin and agent web GUI use this URL:
http://192.168.0.2

username: admin
password: vicidialnow

For access to VtigerCRM use this URL:
http://192.168.0.2/vtigercrm

username: admin
password: admin

For professional support, visit http://www.vicidialnow.com or send an
email to: support@vicidialnow.com

-------------------------------------------------
Don't forget to run update_server_ip everytime you change your IP address

[root@vicidialnow ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2617)
-- Remote UNIX connection
Verbosity is at least 21
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("SIP/cc101-089ecbf8", "agi://127.0.0.1:4577/call_log") in n ew stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-089ecbf8", "SIP/12062036895@sip||tTor") in new stack
-- Called 12062036895@sip
-- Executing NoOp("SIP/66.54.140.46-089ff680", "from-ipkall") in new stack
-- Executing NoOp("SIP/66.54.140.46-089ff680", "INSPIRED MKTG/2064949182") i n new stack
-- Executing Dial("SIP/66.54.140.46-089ff680", "Local/200 at internal") in n ew stack
-- Called 200 at internal
-- Executing AGI("Local/200 at internal@default-6028,2", "agi://127.0.0.1:45 77/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/200 at internal@default-6028,2", "SIP/200 at intern al@sip||tTor") in new stack
-- Called 200 at internal@sip
-- Local/200 at internal@default-6028,1 is ringing
Jan 14 00:53:24 ERROR[2629]: chan_sip.c:11355 handle_request: Missing Cseq. Drop ping this SIP message, it's incomplete.
-- SIP/sip-089f71c8 is making progress passing it to SIP/cc101-089ecbf8
== Refreshing DNS lookups.
== Refreshing DNS lookups.
Jan 14 00:53:32 ERROR[2629]: chan_sip.c:11355 handle_request: Missing Cseq. Drop ping this SIP message, it's incomplete.
== Spawn extension (default, 12062036895, 2) exited non-zero on 'SIP/cc101-089 ecbf8'
-- Executing DeadAGI("SIP/cc101-089ecbf8", "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 DeadAGI("SIP/cc101-089ecbf8", "agi://127.0.0.1:4577/VD_hangup-- HVcauses--PRI-----NODEBUG-----0-----CANCEL----------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... DEBUG----- 0-----CANCEL----------) completed, returning 0
== Spawn extension (from-ipkall, 901835, 3) exited non-zero on 'SIP/66.54.140. 46-089ff680'
== Spawn extension (default, 200 at internal, 2) exited non-zero on 'Local/200 at internal@default-6028,2'
-- Executing DeadAGI("Local/200 at internal@default-6028,2", "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 DeadAGI("Local/200 at internal@default-6028,2", "agi://127.0.0. 1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... DEBUG----- 0-----CANCEL----------) completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc101-089f71c8 was answered.
-- Executing MeetMe("SIP/cc101-089f71c8", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing NoOp("SIP/66.54.140.46-089f4320", "from-ipkall") in new stack
-- Executing NoOp("SIP/66.54.140.46-089f4320", "INSPIRED MKTG/2064949182") in n ew stack
-- Executing Dial("SIP/66.54.140.46-089f4320", "Local/200 at internal") in new stack
-- Called 200 at internal
-- Executing AGI("Local/200 at internal@default-5af0,2", "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/200 at internal@default-5af0,2", "SIP/200 at internal@ sip||tTor") in new stack
-- Called 200 at internal@sip
-- Local/200 at internal@default-5af0,1 is ringing
Jan 14 00:54:50 ERROR[2629]: chan_sip.c:11355 handle_request: Missing Cseq. Droppin g this SIP message, it's incomplete.
Jan 14 00:54:58 ERROR[2629]: chan_sip.c:11355 handle_request: Missing Cseq. Droppin g this SIP message, it's incomplete.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (from-ipkall, 901835, 3) exited non-zero on 'SIP/66.54.140.46- 089f4320'
== Spawn extension (default, 200 at internal, 2) exited non-zero on 'Local/200 at internal@default-5af0,2'
-- Executing DeadAGI("Local/200 at internal@default-5af0,2", "agi://127.0.0.1:4 577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/200 at internal@default-5af0,2", "agi://127.0.0.1:4 577/VD_hangup--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... UG-----0-- ---CANCEL----------) completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
vicidialnow*CLI>

Another interesting thing to notice is that my DID si 2062036895 but CLI shows the same as

("INSPIRED MKTG/2064949182")

Also, if my entry in extension.conf file is

[from-ipkall]
exten => 901835,1,NoOp(from-ipkall)
exten => 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => 901835,3,Dial(Local/200 at internal)

Call lands up in Asterisk but if entry in extension.conf is

[from-ipkall]
exten => 2062036895,1,Ringing()
exten => 2062036895,2,Wait(1)
exten => 2062036895,3,Answer()
exten => 2062036895,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----2062036895-----Closer---------------999-----1)
exten => 2062036895,5,Hangup()

Call DOESNT land up.

by **mflorell** » Wed Jan 14, 2009 10:56 am

What version of VICIDIAL are you using?

by **kolucoms6** » Wed Jan 14, 2009 10:57 am

VERSION: 2.0.4-121 BUILD: 80424-0442

by **mflorell** » Wed Jan 14, 2009 11:00 am

That AGI is wrong, you should be using the _ALL_inbound AGI with that version.

by **kolucoms6** » Wed Jan 14, 2009 11:02 am

after that, here is a sample of the setup for the inbound in extensions.conf in [default]:

exten => 4075060000,1,Ringing()
exten => 4075060000,2,Wait(1)
exten => 4075060000,3,Answer()
exten => 4075060000,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----4075060000-----Closer---------------999-----1)
exten => 4075060000,5,Hangup()

After that you create "SALESLINE" in "ingroups" to link this dialplan entry to the programming in Vicidial.

William suggested that and thats why I copied it.

Let me modify it and get back to you with updates.

by **kolucoms6** » Wed Jan 14, 2009 1:33 pm

When I use

[from-ipkall]
exten => 901835,1,NoOp(from-ipkall)
exten => 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => 901835,3,Dial(Local/200 at internal)

I get below CLI :

login as: root
root@192.168.0.2's password:
Last login: Wed Jan 14 13:09:59 2009 from 192.168.0.21

Welcome to VICIDIALNOW!!!
-------------------------------------------------

For access to the VICIDIAL admin and agent web GUI use this URL:
http://192.168.0.2

username: admin
password: vicidialnow

For access to VtigerCRM use this URL:
http://192.168.0.2/vtigercrm

username: admin
password: admin

For professional support, visit http://www.vicidialnow.com or send an
email to: support@vicidialnow.com

-------------------------------------------------
Don't forget to run update_server_ip everytime you change your IP address

[root@vicidialnow ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2642)
Verbosity is at least 21
-- Executing NoOp("SIP/66.54.140.46-091c1d68", "from-ipkall") in new stack
-- Executing NoOp("SIP/66.54.140.46-091c1d68", "INSPIRED MKTG/2064949182") in new stack
-- Executing Dial("SIP/66.54.140.46-091c1d68", "Local/200 at internal") in new stack
-- Called 200 at internal
-- Executing AGI("Local/200 at internal@default-6781,2", "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/200 at internal@default-6781,2", "SIP/200 at internal@sip||tTor") in new stack
-- Called 200 at internal@sip
-- Local/200 at internal@default-6781,1 is ringing
Jan 14 13:29:25 ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete.
== Spawn extension (from-ipkall, 901835, 3) exited non-zero on 'SIP/66.54.140.46-091c1d68'
== Spawn extension (default, 200 at internal, 2) exited non-zero on 'Local/200 at internal@default-6781,2'
-- Executing DeadAGI("Local/200 at internal@default-6781,2", "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 DeadAGI("Local/200 at internal@default-6781,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
Jan 14 13:29:33 ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete.
vicidialnow*CLI>

When I use

exten => 20620368XX,1,Ringing ; call ringing
exten => 20620368XX,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 20620368XX,3,Answer ; Answer the line
exten => 20620368XX,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-----LB-----SALESLINE-----20620368XX-----Closer-----park----------999-----1-----TESTCAMP)
exten => 20620368XX,5,Hangup

I get engage tone.

Any help ?

by **kolucoms6** » Wed Jan 14, 2009 6:05 pm

Tried to take out the Log file agiout but there is no entry for Inbound keyword.

I am at loss.. Kindly guide me

vicidial.org

Inbnd Configuration

Inbnd Configuration

inbound

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