Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the thank you message.
I started investigating and found that that happens when the call gets
transferred to an agent who is making an outbound call (either calling
customers or logging out). The debug logs of one such conversation is
given below:
As you can read below, the call gets fwd to agent 1005 at SIP/1004. But
he is trying to log off at the same time, and call gets disconnected.
Any help to fix this will be very much appreciated.
regards,
raj
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Goto("Zap/2-1", "MainMenu|s|1") in new
stack
-- Goto (MainMenu,s,1)
-- Executing BackGround("Zap/2-1", "Welcome") in new
stack
-- Playing 'Welcome' (language 'en')
-- Playing 'agent-incorrect' (language 'en')
== Spawn extension (from-sip, 2002, 1) exited non-zero on
'SIP/1004-cc2f'
-- Executing Queue("Zap/2-1", "callcenter|tT|||300") in
new stack
-- Started music on hold, class 'default', on Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Playing 'queue-youarenext' (language 'en')
-- Executing AgentCallbackLogin("SIP/1004-e376", "|l")
in new stack
-- Playing 'agent-user' (language 'en')
-- Told Zap/2-1 in callcenter their queue position (which was 1)
-- Playing 'queue-thankyou' (language 'en')
-- Started music on hold, class 'default', on Zap/2-1
-- outgoing agentcall, to agent '1005', on
'Local/1004@from-sip-d281,1'
-- Executing Dial("Local/1004@from-sip-d281,2",
"SIP/1004") in new
stack
Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call
from user '1004' rejected due to usage limit of 1
-- Couldn't call 1004
== Everyone is busy/congested at this time
-- Called Agent/1005
-- Playing 'agent-incorrect' (language 'en')
== Spawn extension (from-sip, 2002, 1) exited non-zero on
'SIP/1004-e376'
-- Timeout on Local/1004@from-sip-d281,2
== CDR updated on Local/1004@from-sip-d281,2
-- Executing BackGround("Local/1004@from-sip-d281,2",
"vm-goodbye")
in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Agent/1005 answered Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Executing Hangup("Local/1004@from-sip-d281,2", "")
in new stack
== Spawn extension (from-sip, t, 2) exited non-zero on
'Local/1004@from-sip-d281,2'
monitor executing ( nice -n 19 soxmix
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-in.wav"
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-out.wav"
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010.wav"
&& rm -f
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-"*
) &
== Spawn extension (MainMenu, s, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
sip.conf entry for the phone is
[1004]
host=dynamic
type=friend
dtmfmode=RFC2833
username=1004
secret=password
context = from-sip
disallow=all
allow=speex
allow=gsm
incominglimit=1
Bump! raj Rajkumar S wrote:> Hi, > > I have a small call center with 4 Zap lines and 4 agents. Agents login > using sip phones with AgentCallbackLogin. I occasionally gets a > complaint that when customers call the call center, after the initial > greeting is over the call gets cut after playing the thank you message. > I started investigating and found that that happens when the call gets > transferred to an agent who is making an outbound call (either calling > customers or logging out). The debug logs of one such conversation is > given below: > > As you can read below, the call gets fwd to agent 1005 at SIP/1004. But > he is trying to log off at the same time, and call gets disconnected. > > Any help to fix this will be very much appreciated. > > regards, > > raj > > -- Executing Answer("Zap/2-1", "") in new stack > -- Executing Goto("Zap/2-1", "MainMenu|s|1") in new stack > -- Goto (MainMenu,s,1) > -- Executing BackGround("Zap/2-1", "Welcome") in new stack > -- Playing 'Welcome' (language 'en') > -- Playing 'agent-incorrect' (language 'en') > == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-cc2f' > -- Executing Queue("Zap/2-1", "callcenter|tT|||300") in new stack > -- Started music on hold, class 'default', on Zap/2-1 > -- Stopped music on hold on Zap/2-1 > -- Playing 'queue-youarenext' (language 'en') > -- Executing AgentCallbackLogin("SIP/1004-e376", "|l") in new stack > -- Playing 'agent-user' (language 'en') > -- Told Zap/2-1 in callcenter their queue position (which was 1) > -- Playing 'queue-thankyou' (language 'en') > -- Started music on hold, class 'default', on Zap/2-1 > -- outgoing agentcall, to agent '1005', on 'Local/1004@from-sip-d281,1' > -- Executing Dial("Local/1004@from-sip-d281,2", "SIP/1004") in new > stack > Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call > from user '1004' rejected due to usage limit of 1 > -- Couldn't call 1004 > == Everyone is busy/congested at this time > -- Called Agent/1005 > -- Playing 'agent-incorrect' (language 'en') > == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-e376' > -- Timeout on Local/1004@from-sip-d281,2 > == CDR updated on Local/1004@from-sip-d281,2 > -- Executing BackGround("Local/1004@from-sip-d281,2", "vm-goodbye") > in new stack > -- Playing 'vm-goodbye' (language 'en') > -- Agent/1005 answered Zap/2-1 > -- Stopped music on hold on Zap/2-1 > -- Executing Hangup("Local/1004@from-sip-d281,2", "") in new stack > == Spawn extension (from-sip, t, 2) exited non-zero on > 'Local/1004@from-sip-d281,2' > monitor executing ( nice -n 19 soxmix > "/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-in.wav" > "/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-out.wav" > "/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010.wav" > && rm -f > "/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-"* > ) & > == Spawn extension (MainMenu, s, 2) exited non-zero on 'Zap/2-1' > -- Hungup 'Zap/2-1' > > sip.conf entry for the phone is > > [1004] > host=dynamic > type=friend > dtmfmode=RFC2833 > username=1004 > secret=password > context = from-sip > disallow=all > allow=speex > allow=gsm > incominglimit=1 >