How can we help you today?
Start a new topic

Transfer not allowed when doing outgoing calls

Hi everyone,

we have an issue here which my customers are getting mad at me, so i need to solve this very fast :)


When my customer has an OUTGOING Call on his Snom D385, he isnt able to transfer this call to another one of his other Snom D385s.


Every time he trys, doesnt matter if blind or attended, it issues "Transfer not allowed".


When getting incoming calls, every transfer is working normal.


Does anybody know what it could be?


PS:


My customer just asked me if it isnt possible to have only ONE quick transfer button for every other snom. Before, they had Grandstream IP Phones and when there was a call, you could press a button and either transfer directly blind via hanging up or transfer attended while waiting for the other party to answer.

ATM he has one blind transfer button and uses the BLF button together with two times transfer button für attended transfer. But its not very comfortable.

I already looked at the wiki, but for me its not very helpful, even being an it-technician myself, sorry. -.-


Marcus,


Which phone and what firmware version? Have you looked at the SIP trace to see if the hold INVITE or REFER from the D385 to the transferee or transfer target resulted in a 2xx response? For example, here is the SIP signaling for an Attended Call Transfer


Attended Transfer

SIP Call Flow


A
SIP Proxy (SER)
B
SIP Proxy (SER)
C
#1 A initiates call to B--- INVITE SDP ---> | --- INVITE SDP --->

<--- 100 trying --- |
#2 B is ringing<--- 180 Ringing --- | <--- 180 Ringing ---
#3 B accepts the call<--- 200 SDP OK --- | <--- 200 SDP OK ---
#4 A acknowledges--- ACK ---> | --- ACK --->
#5 A and B talk<--- RTP (Audio) --->
#6 B places A on Hold<--- INVITE SDP --- | <--- INVITE SDP ---
#7 A accepts Hold--- 200 SDP OK ---> | --- 200 SDP OK --->
#8 B acknowledges<--- ACK --- | <--- ACK ---
#9 B initiates call to C
--- INVITE SDP ---> | --- INVITE SDP --->
#10 C is ringing
<--- 100 Trying --- |
#11 C is ringing
<--- 180 Ringing --- | <--- 180 Ringing ---
#12 C accepts the call
<--- 200 SDP OK --- | <--- 200 SDP OK ---
#13 B acknowledges
--- ACK ---> | --- ACK --->
#14 B announces transfer to A
<--- RTP (Audio) --->
#15 B presses "transfer" and C is automatically placed on hold
--- INVITE SDP ---> | --- INVITE SDP --->
#16 placing C on hold
<--- 100 Trying --- |
#17 C accepts hold
<--- 200 SDP OK --- | <--- 200 SDP OK ---
#18 B acknowledges
--- ACK ---> | --- ACK --->
#19 B presses "transfer" again and refers A to C<--- REFER --- | <--- REFER ---
#20 A accepts the next call goes to C--- 202 Accepted ---> | --- 202 Accepted --->
#21 A initiates call to C--- INVITE SDP ---> | --- INVITE SDP ----------------------------------------------------------------------------->
#22 C accepts the call from A<--- 200 SDP OK --- | <--- 200 SDP OK ------------------------------------------------------------------------------
#23 C terminates B call
<--- BYE --- | <--- BYE ---
#24 B closes C call
--- OK ---> | --- OK --->
#25 A informs B to terminate call--- NOTIFY ---> | --- NOTIFY --->
#26 A acknowledges--- ACK ---> | --- ACK ------------------------------------------------------------------------------------------------->
#27 A talks to C<---------------------------------------------------- RTP (Audio) ---------------------------------------------------->
#28 B receives NOTIFY<--- OK --- | <--- OK ---
#29 B terminates A call<--- BYE --- | <--- BYE ---
#30 A closes B call--- OK ---> | --- OK --->


The SIP trace should provide more guidance.


You can use the Smart Transfer options in the new version 10 firmware to select Transfer > Pool > Monitored Extensions (BLF) transfer target.


https://service.snom.com/display/wiki/Smart+Transfer+-+V10



Regards,


Snom Support


Thanks for you response.

We got five D385 with the latest firmware (installed it yesterday to test if this changes anything, but didnt)


As i have learned the hard way, it seems like a lot of sip providers do not support reinvites or cut their headers and this could be the problem.


We are using sipgate.de and it doesnt work there. I tried placetel a few minutes ago - the same..


I really hoped that i could find a solution but it doesnt seem like i will find one.


I will send a sip trace later.

Login or Signup to post a comment