thr3ads.net - asterisk users - [Asterisk-Users] transferring a IAX call into a conference [Feb 2005]

If this information is useful, please help other people find it:
Share via:

dean collins

2005-Feb-11 06:33 UTC

[Asterisk-Users] transferring a IAX call into a conference

When I make a call out on the Faktortel number I am then able to
transfer to call to my asterisk meetme room of 801 by hitting 'transfer'
then '801' then 'send' on my grandstream phone.

 

This connects my faktortel trunk (and who ever is on the other end) to
my conference room I can then make another call using my local pstn
service and set up a 3 way (or whatever number in a conference call)

 

 

 

Now the problem is this;

 

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.

 

 

 

Cheers,

Dean

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050211/93233879/attachment.htm

timebandit001@gmail.com

2005-Feb-11 06:45 UTC

head link

[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 :)

dean collins

2005-Feb-11 07:18 UTC

head link

[Asterisk-Users] transferring a IAX call into a conference

I'm using an asterisk@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@Faktortel-out/5",
"0?from-pstn-reghours|s|1:") in new stack
    -- Executing GotoIf("IAX2/faktortel@Faktortel-out/5",
"0?from-pstn-afthours|s|1:") in new stack
    -- Executing GotoIfTime("IAX2/faktortel@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@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@Faktortel-out/5",
"intype=GRP-700") in new stack
    -- Executing Cut("IAX2/faktortel@Faktortel-out/5",
"intype=intype|-|1") in new stack
    -- Executing GotoIf("IAX2/faktortel@Faktortel-out/5",
"0?4:5") in
new stack
    -- Goto (from-pstn-reghours-nofax,s,5)
    -- Executing GotoIf("IAX2/faktortel@Faktortel-out/5",
"1?6:7") in
new stack
    -- Goto (from-pstn-reghours-nofax,s,6)
    -- Executing Goto("IAX2/faktortel@Faktortel-out/5",
"ext-group|700|1") in new stack
    -- Goto (ext-group,700,1)
    -- Executing SetVar("IAX2/faktortel@Faktortel-out/5",
"GROUP=30|32|33|") in new stack
    -- Executing SetVar("IAX2/faktortel@Faktortel-out/5",
"RINGTIMER=30") in new stack
    -- Executing SetVar("IAX2/faktortel@Faktortel-out/5",
"PRE=4357") in
new stack
    -- Executing Macro("IAX2/faktortel@Faktortel-out/5",
"rg-group") in
new stack
    -- Executing SetVar("IAX2/faktortel@Faktortel-out/5",
"GRP=30|32|33|") in new stack
    -- Executing SetGroup("IAX2/faktortel@Faktortel-out/5",
"") in new
stack
    -- Executing SetVar("IAX2/faktortel@Faktortel-out/5",
"FROMCONTEXT=rg-group") in new stack
    -- Executing SetCIDName("IAX2/faktortel@Faktortel-out/5",
"4357") in
new stack
    -- Executing Macro("IAX2/faktortel@Faktortel-out/5",
"dial|30|tr|30|32|33|") in new stack
    -- Executing AGI("IAX2/faktortel@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@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@Faktortel-out/5
    -- Started music on hold, class 'default', on
IAX2/faktortel@Faktortel-out/5
    -- Stopped music on hold on IAX2/faktortel@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@Faktortel-out/5' jumping out of macro
'dial'
  == Channel 'IAX2/faktortel@Faktortel-out/5' jumping out of macro
'rg-group'
    -- Executing Macro("IAX2/faktortel@Faktortel-out/5",
"hangupcall")
in new stack
    -- Executing ResetCDR("IAX2/faktortel@Faktortel-out/5",
"w") in new
stack
    -- Executing NoCDR("IAX2/faktortel@Faktortel-out/5", "")
in new
stack
    -- Executing Wait("IAX2/faktortel@Faktortel-out/5", "5")
in new
stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'IAX2/faktortel@Faktortel-out/5' in macro 'hangupcall'
  == Spawn extension (from-internal, s, 1) exited non-zero on
'IAX2/faktortel@Faktortel-out/5'
    -- Hungup 'IAX2/faktortel@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
timebandit001@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 :)
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@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

Possibly Parallel Threads

Search for more reasonably related threads

asterisk users - Feb 2005 - transferring a IAX call into a conference

[Asterisk-Users] transferring a IAX call into a conference

[Asterisk-Users] transferring a IAX call into a conference

[Asterisk-Users] transferring a IAX call into a conference

Possibly Parallel Threads