[Asterisk-Users] transferring a IAX call into a conference
dean collins
dean at collins.net.pr
Fri Feb 11 07:18:16 MST 2005
I'm using an asterisk at home installation.
I dialed out on my packet8 service using a '9'
And dialed back in my faktortel iax service.
I have tried this with people dialing into my Faktortel service as well
using my cell phone but same thing happens.
asterisk1*CLI>
asterisk1*CLI>
asterisk1*CLI>
-- Executing Macro("SIP/30-e7e2", "dialout|1|961283073503") in new
stack
-- Executing SetVar("SIP/30-e7e2", "length=1") in new stack
-- Executing Dial("SIP/30-e7e2", "ZAP/g0/61283073503") in new stack
-- Called g0/61283073503
-- Zap/1-1 answered SIP/30-e7e2
-- Accepting AUTHENTICATED call from 202.125.42.141, requested
format = 256, actual format = 1024
-- Executing GotoIf("IAX2/faktortel at Faktortel-out/5",
"0?from-pstn-reghours|s|1:") in new stack
-- Executing GotoIf("IAX2/faktortel at Faktortel-out/5",
"0?from-pstn-afthours|s|1:") in new stack
-- Executing GotoIfTime("IAX2/faktortel at Faktortel-out/5",
"5:55-23:59|*|*|*?from-pstn-reghours|s|1:") in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("IAX2/faktortel at Faktortel-out/5",
"1?from-pstn-reghours-nofax|s|1:2") in new stack
-- Goto (from-pstn-reghours-nofax,s,1)
-- Executing SetVar("IAX2/faktortel at Faktortel-out/5",
"intype=GRP-700") in new stack
-- Executing Cut("IAX2/faktortel at Faktortel-out/5",
"intype=intype|-|1") in new stack
-- Executing GotoIf("IAX2/faktortel at Faktortel-out/5", "0?4:5") in
new stack
-- Goto (from-pstn-reghours-nofax,s,5)
-- Executing GotoIf("IAX2/faktortel at Faktortel-out/5", "1?6:7") in
new stack
-- Goto (from-pstn-reghours-nofax,s,6)
-- Executing Goto("IAX2/faktortel at Faktortel-out/5",
"ext-group|700|1") in new stack
-- Goto (ext-group,700,1)
-- Executing SetVar("IAX2/faktortel at Faktortel-out/5",
"GROUP=30|32|33|") in new stack
-- Executing SetVar("IAX2/faktortel at Faktortel-out/5",
"RINGTIMER=30") in new stack
-- Executing SetVar("IAX2/faktortel at Faktortel-out/5", "PRE=4357") in
new stack
-- Executing Macro("IAX2/faktortel at Faktortel-out/5", "rg-group") in
new stack
-- Executing SetVar("IAX2/faktortel at Faktortel-out/5",
"GRP=30|32|33|") in new stack
-- Executing SetGroup("IAX2/faktortel at Faktortel-out/5", "") in new
stack
-- Executing SetVar("IAX2/faktortel at Faktortel-out/5",
"FROMCONTEXT=rg-group") in new stack
-- Executing SetCIDName("IAX2/faktortel at Faktortel-out/5", "4357") in
new stack
-- Executing Macro("IAX2/faktortel at Faktortel-out/5",
"dial|30|tr|30|32|33|") in new stack
-- Executing AGI("IAX2/faktortel at Faktortel-out/5",
"dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 1
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1108130997.18
-- dialparties.agi: channel = IAX2/faktortel at Faktortel-out/5
-- dialparties.agi: callerid = 4357
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = IAX2
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID is not set
-- dialparties.agi: Added extension 30 to extension map
-- dialparties.agi: Added extension 32 to extension map
-- dialparties.agi: Added extension 33 to extension map
-- dialparties.agi: Extension 33 cf is disabled
-- dialparties.agi: Extension 32 cf is disabled
-- dialparties.agi: Extension 30 cf is disabled
-- dialparties.agi: Extension 33 do not disturb is disabled
-- dialparties.agi: Extension 32 do not disturb is disabled
-- dialparties.agi: Extension 30 do not disturb is disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 33 has call waiting disabled
dialparties.agi: Extension 32 has call waiting disabled
dialparties.agi: Extension 30 has call waiting disabled
dialparties.agi: Max calls of 1 exceeded - deleting from dial
dialparties.agi: Dial still has extensions - continuing
-- dialparties.agi: DbDel CALLTRACE/33 - Caller ID is not defined
-- dialparties.agi: DbDel CALLTRACE/32 - Caller ID is not defined
dialparties.agi: About to execute Dial(IAX2/33&SIP/32|30|tr)
-- AGI Script Executing Application: (Dial) Options:
(IAX2/33&SIP/32|30|tr)
-- Called 32
-- SIP/32-30dc is ringing
-- SIP/32-30dc answered IAX2/faktortel at Faktortel-out/5
-- Started music on hold, class 'default', on
IAX2/faktortel at Faktortel-out/5
-- Stopped music on hold on IAX2/faktortel at Faktortel-out/5
dialparties.agi: Dial return value was -1 and dialstring was
IAX2/33&SIP/32|30|tr
dialparties.agi: Setting Priority to 22 from 2
-- AGI Script dialparties.agi completed, returning 0
== Channel 'IAX2/faktortel at Faktortel-out/5' jumping out of macro
'dial'
== Channel 'IAX2/faktortel at Faktortel-out/5' jumping out of macro
'rg-group'
-- Executing Macro("IAX2/faktortel at Faktortel-out/5", "hangupcall")
in new stack
-- Executing ResetCDR("IAX2/faktortel at Faktortel-out/5", "w") in new
stack
-- Executing NoCDR("IAX2/faktortel at Faktortel-out/5", "") in new
stack
-- Executing Wait("IAX2/faktortel at Faktortel-out/5", "5") in new
stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'IAX2/faktortel at Faktortel-out/5' in macro 'hangupcall'
== Spawn extension (from-internal, s, 1) exited non-zero on
'IAX2/faktortel at Faktortel-out/5'
-- Hungup 'IAX2/faktortel at Faktortel-out/5'
-- Hungup 'Zap/1-1'
== Spawn extension (macro-dialout, s, 2) exited non-zero on
'SIP/30-e7e2' in macro 'dialout'
== Spawn extension (from-internal, 961283073503, 1) exited non-zero on
'SIP/30-e7e2'
-- Executing Macro("SIP/30-e7e2", "hangupcall") in new stack
-- Executing ResetCDR("SIP/30-e7e2", "w") in new stack
-- Executing NoCDR("SIP/30-e7e2", "") in new stack
-- Executing Wait("SIP/30-e7e2", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/30-e7e2' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/30-e7e2'
asterisk1*CLI>
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
timebandit001 at gmail.com
Sent: Friday, February 11, 2005 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] transferring a IAX call into a conference
> If someone calls me in on my faktortel number I cant transfer them to
the
> conference call room. It literally disconnects them each time I
transfer?
>
> Why is this? What can I do to prevent this.
Any CLI log from when you try that ?
Help us helping you :)
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