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Summary:ASTERISK-08968: Blind transfers are parsed by [default], but attended transfers work as expected
Reporter:Leif Madsen (lmadsen)Labels:
Date Opened:2007-03-08 12:20:01.000-0600Date Closed:2011-06-07 14:07:24
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Transfers
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) attended_transfer_completes.txt
( 1) blind_transfer_fails.txt
( 2) failed_blind_transfer_w_sip_debug-1.txt
( 3) start_attended_then_transfer_during_ringing_completes.txt
Description:When testing transfers in 1.4, I was getting my Asterisk box trying to parse the transfer in the [default] context instead of the context the phone should be using.

I have a bit of a unique setup which I will try to explain here.

Here is the topology:

       101
        |
102 --- SER --- 100
        |
     Asterisk


The call flow generally does this:

102 --> SER --> Asterisk --> SER --> 100


And the steps I perform:

Working scenario:
1) 102 calls 100
2) 100 Answers
3) 102 attended transfers to 101
4) 101 Answers
5) 102 hits transfer
6) Transfer completes

Failed scenario:
1) 102 calls 100
2) 102 blind transfers to 101
3) 100 hears "no-route-exists-to-dest" <-- parsed via [default] context
4) call dropped

Partially Working scenario:
1) 102 calls 100
2) 102 attended transfers to 101
3) MoH is played
4) while ringing transfer button is hit
5) 102 hangs up
6) MoH stops playing
7) 101 continues to ring
8) 101 answers
9) Call sets up and talking occurs!

****** ADDITIONAL INFORMATION ******

I've attached 3 files which shows what is happening. Yes... it is a lot of dialplan logic... but its near the end of each of the files you want to see.
Comments:By: Leif Madsen (lmadsen) 2007-03-08 12:25:16.000-0600

I've made this private because it is a partial production system, and I didn't want to confuse poor oej by changing things around in the file.

By: Serge Vecher (serge-v) 2007-03-08 12:28:15.000-0600

leif, looks like you've forgotten to enter 'sip set debug'. Oej will want to see that. Also, trimming startup info will cut the file sizes in about a half.

By: Leif Madsen (lmadsen) 2007-03-28 13:39:35

I was basically able to make all of this work with standard stuff in Asterisk, so no change to the code is required here.