kwem at gmx.de
Guest
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 Posted: Tue Apr 15, 2008 12:38 pm Post subject: [asterisk-users] Problem with SIP, attended transfer and GRO |
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Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time. In case
of attended transfer (on SIP-basis, not via the #-mechanism of asterisk)
I have problems.
To simplfy the scenario I stripped down the dialplan to the following.
| Quote: | From somewhere on the wiki I am using the following context:
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exten => 200,1,Set(GROUP()=${CALLERID(num)})
exten => 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} >= 1]?BLOCK)
exten => 200,n,Set(OUTBOUND_GROUP=${EXTEN})
exten => 200,n,Dial(SIP/katrin)
exten => 200,n(BLOCK),Busy
This block is used for other extensions 100 and 150 respectivily. It
works fine until I am using attended transfer.
Example: kwe (Extension 100) is calling katrin (Extension 200). katrin
sets the call on hold and talks to hans (Extension 150).
At the cli I get the following result:
pbxtest*CLI> group show channels
Channel Group Category
SIP/kwe-081bf188 100 (default)
SIP/katrin-081b70a8 200 (default)
SIP/katrin-081bb020 200 (default)
SIP/hans-0816b8b8 150 (default)
which seems correct to me.
In case of a transfer of kwe to hans (katrin leaving), the result is:
pbxtest*CLI> group show channels
Channel Group Category
SIP/kwe-081bf188 100 (default)
SIP/kwe-081bf188 200 (default)
SIP/hans-0816b8b8 150 (default)
I am confused about the second line, which leads to trouble. The above
context would think, that katrin is busy. In case of a blind transfer
everything is ok (the second line does not exist)
I have tested the above with * 1.4.14, 1.4.18-rc4 and 1.4.19
Is this a bug or a feature? Am I doing something wrong or should I file
a bug report?
Thanks in advance,
Regards
Karsten |
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