Summary:ASTERISK-08127: Call transfer or parking failure
Reporter:tootai (tootai)Labels:
Date Opened:2006-11-13 11:58:33.000-0600Date Closed:2006-11-20 11:27:53.000-0600
Versions:Frequency of
Description:I have one SIP phone -Tiger IP202- and one ATA -Sipura 1001-. I call from SIP phone to ATA, answer, and then park or transfer the call from ATA: it failed with following info (debug=5 verbose=5)
-- Started music on hold, class 'default', on SIP/107-082454d0
   -- Playing 'pbx-transfer' (language 'en')
   -- Unable to find extension '13' in context ''
   -- Playing 'pbx-invalid' (language 'en')
   -- Stopped music on hold on SIP/107-082454d0

Parking or transfering the call from SIP phone using the transfer key is ok. Sipura has DTMF mode AVT, the phone has rfc2833 and sip.conf has dtmfmode=auto.  Same behaviour if I put rfc2833.

Now I call from ATA to SIP phone and do the park or transfer from ATA, it's working.

Dial OPT cde is rTt. 13 are the first digit of extensions I want to transfer -which is 134- context is '' but I don't know why and seems that this is the problem.

The calls are done with the same context. I hear musiconhold on the device which will be transfered, as well as the pbx-transfer and pbx-invalid audio on the ATA.
Comments:By: Joshua C. Colp (jcolp) 2006-11-15 18:35:34.000-0600

We need to know the dial string being used, and which side is doing the transfer.

By: tootai (tootai) 2006-11-16 02:15:36.000-0600

The dial string is:

exten => _1[01]X,1,Macro(dialSIP-local,${EXTEN},${DIALOPT})

Where DIALOPT is a global variables which contains rTt. The macro is


exten => s,1,Dial(SIP/${ARG1},${SIPTIMERING},${ARG2})
exten => s,2,Goto(s-${DIALSTATUS},1)

where SIPTIMERING contains 60. The result is:

Dial("SIP/107-0825d5d8", "SIP/106|60|rTt") in new stack
   -- Called 106

107 being the IP phone, 106 the Sipura.

We have this behaviour when the Sipura is trying to do the transfer. Remember that if Sipura is calling the IP phone and is doing the transfer, it works.

By: Serge Vecher (serge-v) 2006-11-20 10:51:07.000-0600

Since you are using SIP transfer, please also include the sip debug, logged as per following:

1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddgc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Attach verbosedebug.txt to the issue.

By: Joshua C. Colp (jcolp) 2006-11-20 11:27:53.000-0600

Should be fixed in 1.4 as of revision 47850 and trunk as of revision 47851. If it's still an issue, then reopen.