[Asterisk-Users] AgentLogin issue
Joe Dennick
joe at dennick.net
Thu Aug 12 20:46:34 MST 2004
Did you also define the agents in agents.conf?
It should look like this:
<agents.conf>
; Agent configuration
;
;
[agents]
autologoff=14
ackcall=no
wrapuptime=5000
musiconhold => default
agent => 1001,1001,Agent1
agent => 1002,1002,Agent2
<end agents.conf>
Please note that the proper syntax is 'agent => agent-id, password,
agent-name.
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Maurizio
Marini
Sent: Thursday, August 12, 2004 9:50 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member => Agent/1001
member => Agent/1002
extension.conf
exten => 110,1,Wait,1
exten => 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route:
build_route: Contact hop: <sip:sip3 at 192.168.1.151:5060>
-- Executing Wait("SIP/sip3-768a", "1") in new stack
-- Executing AgentLogin("SIP/sip3-768a", "") in new stack Aug 12
16:31:37 DEBUG[1127562160]: rtp.c:1156 ast_rtp_write: Ooh, format
changed from UNKN to ULAW Aug 12 16:31:37 DEBUG[1127562160]:
channel.c:1101 ast_settimeout: Scheduling timer at 160 sample intervals
-- Playing 'agent-user' (language 'en')
Aug 12 16:31:37 DEBUG[1103408048]: chan_sip.c:817 __sip_ack: Stopping
retransmission on '78383678327d335d' of Response 2: Found Aug 12
16:31:41 DEBUG[1127562160]: channel.c:1101 ast_settimeout: Scheduling
timer at 0 sample intervals Aug 12 16:31:41 DEBUG[1127562160]:
channel.c:1101 ast_settimeout: Scheduling timer at 0 sample intervals
Aug 12 16:31:42 DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 49
(1), at 192.168.1.151 Aug 12 16:31:43 DEBUG[1127562160]: rtp.c:189
send_dtmf: Sending dtmf: 48 (0), at 192.168.1.151 Aug 12 16:31:44
DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 48 (0), at
192.168.1.151 Aug 12 16:31:46 DEBUG[1127562160]: rtp.c:189 send_dtmf:
Sending dtmf: 49 (1), at 192.168.1.151 Aug 12 16:31:47
DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 35 (#), at
192.168.1.151
== Spawn extension (local, 110, 2) exited non-zero on 'SIP/sip3-768a'
Aug 12 16:31:51 DEBUG[1127562160]: cdr_addon_mysql.c:178 mysql_log:
cdr_mysql: inserting a CDR record. Aug 12 16:31:51 DEBUG[1127562160]:
cdr_addon_mysql.c:197 mysql_log: cdr_mysql: SQL command as follows:
INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES
('2004-08-12 16:31:36','\"sip3\" <103>','103','110','local',
'SIP/sip3-768a','','AgentLogin','',15,14,'ANSWERED',3,'','1092321096.2',
'')
my call is interpreted as a phon call and cdr record it :(
what am i missing?
thnx for help
m.
--
Maurizio Marini GSM +39-335-8259739
Work: +39-0721-855285 Fax +39-0721-859609
Home: +39-0721-950396 IAXTel: (700) 350-1234
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---
Incoming mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004
More information about the asterisk-users
mailing list