Transfer Doesn't seem to work?

Get a call, to Transfer, I press ##, and hear the message, [bold]Trasnfer[/bold], dial an
extension, get the message “I’m sorry, not a valid extension”. CLI shows the following:

– Started music on hold, class ‘default’, on SIP/112.113.114.115-b7b09210
– Playing ‘pbx-transfer’ (language ‘en’)
– Unable to find extension ‘’ in context ‘from-internal-xfer’
– Playing ‘pbx-invalid’ (language ‘en’)
– Stopped music on hold on SIP/112.113.114.115-b7b09210
– Attempting native bridge of SIP/112.113.114.115-b7b09210 and SIP/100-4025

These are the relevant sections from my config files;

extensions.conf:

[from-internal-xfer]
; applications are now mostly all found in from-internal-additional in _custom.conf
include => parkedcalls
include => from-internal-custom
;allow phones to dial other extensions
include => ext-fax
;allow phones to access generated contexts
;
; MODIFIED (PL)
;
; Currently the include for findmefollow is being auto-generated before ext-local which is the desired behavior.
; However, I haven’t been able to do anything that I know of to force this. We need to determine if it should
; be hardcoded into here to make sure it doesn’t change with some configuration. For now I will leave it out
; until we can discuss this.
;
include => ext-local-confirm
include => findmefollow-ringallv2
include => from-internal-additional
; This causes grief with ‘#’ transfers, commenting out for the moment.
; include => bad-number
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

[from-internal]
include => from-internal-xfer
include => bad-number

From globals in extensions_additional.conf:

[globals]
#include globals_custom.conf


;TRANSFER_CONTEXT = from-internal-xfer
TRANSFER_CONTEXT = from-internal ; I chaged this as a trial

Any ideas why this is happening?

Sounds like you are pressing ‘##’ and then nothing else. You need to follow that with the number where you want it transfered. The message shows the number you tried to transfer to in single quotes. It is showing ‘’ or otherwise it did not see a number. If you had provided an invalid number you would see that number in quotes and get the same message.

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

I am having a strange problem with freepbx 2.3.0.3 (asterisk 1.4.10.1), blind transfer has stopped working completely. After I press ## to initiate transfer, it plays the transfer message to me, and starts music on hold for remote party. I enter the extension, and then asterisk drops both ends of the call.

Here is a part of the log file:

[Oct 9 12:35:19] VERBOSE[20823] logger.c: – Started music on hold, class ‘default’, on SIP/811-086a0670
[Oct 9 12:35:19] VERBOSE[20823] logger.c: – Playing ‘pbx-transfer’ (language ‘en’)
[Oct 9 12:35:24] VERBOSE[19652] logger.c:
<— SIP read from 192.168.10.78:1826 —>

<------------->
[Oct 9 12:35:24] VERBOSE[19652] logger.c: — (0 headers 1 lines) —
[Oct 9 12:35:25] VERBOSE[20823] logger.c: – Stopped music on hold on SIP/811-086a0670
[Oct 9 12:35:25] VERBOSE[20823] logger.c: – Transferring SIP/811-086a0670 to ‘100’ (context from-internal-xfer) priority 1
[Oct 9 12:35:25] DEBUG[20823] chan_sip.c: Call to peer ‘812’ removed from call limit 50
[Oct 9 12:35:25] VERBOSE[20823] logger.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 7104 ms (Method: INVITE)
[Oct 9 12:35:25] DEBUG[20823] chan_sip.c: Strict routing enforced for session [email protected]
[Oct 9 12:35:25] VERBOSE[20823] logger.c: set_destination: Parsing for address/port to send to
[Oct 9 12:35:25] VERBOSE[20823] logger.c: set_destination: set destination to 192.168.10.225, port 7790
[Oct 9 12:35:25] VERBOSE[20823] logger.c: Reliably Transmitting (NAT) to 192.168.10.225:7790:
BYE sip:[email protected]:7790;rinstance=479335fb91c79c6e SIP/2.0
Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK6a2869a5;rport
From: “Philip” ;tag=as53c76fd0
To: ;tag=f80cc44b
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


[Oct 9 12:35:25] VERBOSE[20823] logger.c: – fixed jitterbuffer destroyed on channel SIP/812-0863a8c8
[Oct 9 12:35:25] DEBUG[20823] app_macro.c: Executed application: Dial
[Oct 9 12:35:25] VERBOSE[20823] logger.c: == Channel ‘SIP/811-086a0670’ jumping out of macro ‘dial’
[Oct 9 12:35:25] DEBUG[20823] app_macro.c: Executed application: Macro
[Oct 9 12:35:25] VERBOSE[20823] logger.c: == Channel ‘SIP/811-086a0670’ jumping out of macro ‘exten-vm’
[Oct 9 12:35:25] VERBOSE[20823] logger.c: – Executing [[email protected]:1] Macro(“SIP/811-086a0670”, “hangupcall”) in new stack
[Oct 9 12:35:25] VERBOSE[20823] logger.c: – Executing [[email protected]:1] ResetCDR(“SIP/811-086a0670”, “w”) in new stack
[Oct 9 12:35:25] DEBUG[20823] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
[Oct 9 12:35:25] DEBUG[20823] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2007-10-09 12:34:51’,’“Philip” <811>’,‘811’,‘812’,‘from-internal’, ‘SIP/811-086a0670’,‘SIP/812-0863a8c8’,‘Dial’,‘SIP/812||TtrwW’,34,25,‘ANSWERED’,3,’’,‘1191918891.75’)

Has anyone experienced this problem and knows how to fix it?

Thanks in advance!

Same Here… Asterisk is just dumping core for me. Running 2.3.x and Asterisk 1.4.13

Seems to be an issue…

This is a major problem on our 40 user, 200+ incoming calls per day system. We may have to go back to an older version if I cannot find a workaround today

I thought I saw another post on this and it was an Asterisk issue. The question to all of you who are having this problem is simple. Are you running Asterisk 1.4.x because you are taking advantage of something specific in the 1.4 tree that is not available in 1.2. If you are not doing anything beyond FreePBX standard usage then teh answer should be no. If that is the case, then go back to Asterisk 1.2 until Asterisk 1.4 can get through a few more versions. It has come a long way, but the reliability is clearly not at the level of 1.2 yet and if your production systems are being effected, … (Keep in mind, Asterisk Business Edition was still a 1.2 base last I checked about a month ago and I suspect that is still the case, that should tell you something).

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

I am having a similar problem running freepbx 2.3.1 (asterisk 1.4.10.1, actually Trixbox), but only with blind Xfers made by the called party, the caller can execute blind transfers correctly.
Found a workaround by enabling attended transfer, uncommenting the line for feature code *2, that works fine.

Cheers!

Would you happen to remember which post you saw this on? Was it this site or another? I thought I searched fairly well on here, but it is entirely possible I missed something. I would really like to make 1.4.x work because our queues have been more stable since we upgraded.