1.2.16 crashes after transfer with BT-100

Asterisk Asterisk Support
1

Hi :smile:

I have the following call:
Cell -> (VoIP provider) -> Asterisk -> OpenSER -> VoIP phone A

Call works perfectly and now I will transfer the call so I press “flash”, dial the voip number (same net) and the new call works:
VoIP phone A -> OpenSER -> Asterisk -> OpenSer -> VoIP phone B

Now I press “transfer” on the BT-100 (phone A) and Asterisk crashes:

Apr 19 16:41:20 DEBUG[17699]: chan_sip.c:11299 handle_request: **** Received REFER (9) - Command in SIP REFER
Apr 19 16:41:20 DEBUG[17699]: chan_sip.c:10817 handle_request_refer: SIP call transfer received for call 5b9eaedb4e69bb364939108a7cbc0dc3@myasterisk (REFER)!
Apr 19 16:41:20 DEBUG[17699]: chan_sip.c:6973 get_refer_info: Assigning Replace-Call-ID Info 4d7ff52065487d43@myphoneA to REPLACE_CALL_ID
Apr 19 16:41:20 DEBUG[17699]: chan_sip.c:10829 handle_request_refer: 202 Accepted (supervised)
Apr 19 16:41:20 DEBUG[17699]: channel.c:2409 set_format: Set channel SIP/myOpenSER-081b7a78 to write format alaw
Apr 19 16:41:20 DEBUG[17699]: channel.c:2834 ast_channel_masquerade: Planning to masquerade channel SIP/myasterisk-081ab9e0 into the structure of SIP/myOpenSer-081c8260
Apr 19 16:41:20 DEBUG[17699]: channel.c:2847 ast_channel_masquerade: Done planning to masquerade channel SIP/myasterisk-081ab9e0 into the structure of SIP/myOpenSer-081c8260
Apr 19 16:41:20 DEBUG[18002]: channel.c:2961 ast_do_masquerade: Actually Masquerading SIP/myasterisk-081ab9e0(6) into the structure of SIP/myOpenSer-081c8260(6)
Apr 19 16:41:20 DEBUG[18002]: channel.c:2972 ast_do_masquerade: Got clone lock for masquerade on 'SIP/myasterisk-081ab9e0' at 0x81327c4
Apr 19 16:41:20 DEBUG[18002]: chan_sip.c:2462 sip_hangup: Hangup call SIP/myasterisk-081ab9e0<MASQ>, SIP callid 4d7ff52065487d43@myphoneA)
Apr 19 16:41:20 DEBUG[18002]: chan_sip.c:2470 sip_hangup: update_call_counter() - decrement call limit counter
Apr 19 16:41:20 DEBUG[18002]: chan_sip.c:2252 update_call_counter: Updating call counter for incoming call
Apr 19 16:41:20 DEBUG[18002]: channel.c:2409 set_format: Set channel SIP/myasterisk-081ab9e0 to write format slin
Apr 19 16:41:20 DEBUG[18002]: channel.c:2409 set_format: Set channel SIP/myasterisk-081ab9e0 to read format slin
Apr 19 16:41:20 DEBUG[18002]: channel.c:3154 ast_do_masquerade: Putting channel SIP/myasterisk-081ab9e0 in 64/64 formats
Apr 19 16:41:20 DEBUG[18002]: channel.c:3189 ast_do_masquerade: Released clone lock on 'SIP/myOpenSer-081c8260<ZOMBIE>'
Apr 19 16:41:20 DEBUG[18002]: channel.c:3198 ast_do_masquerade: Done Masquerading SIP/myasterisk-081ab9e0 (6)
Apr 19 16:41:20 DEBUG[17790]: channel.c:3377 ast_generic_bridge: Didn't get a frame from channel: SIP/myOpenSer-081c8260<ZOMBIE>
Apr 19 16:41:20 DEBUG[17790]: channel.c:3660 ast_channel_bridge: Bridge stops bridging channels SIP/myOpenSer-081c8260<ZOMBIE> and SIP/myOpenSer-081b7a78
Apr 19 16:41:20 DEBUG[17790]: channel.c:1373 ast_hangup: Hanging up channel 'SIP/myOpenSer-081b7a78'
Apr 19 16:41:20 DEBUG[17790]: chan_sip.c:2462 sip_hangup: Hangup call SIP/myOpenSer-081b7a78, SIP callid 5b9eaedb4e69bb364939108a7cbc0dc3@myOpenser)
Apr 19 16:41:20 DEBUG[17790]: chan_sip.c:2470 sip_hangup: update_call_counter(49180xxxxxxxxxx) - decrement call limit counter
Apr 19 16:41:20 DEBUG[17790]: chan_sip.c:2252 update_call_counter: Updating call counter for incoming call
Apr 19 16:41:20 DEBUG[17790]: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
Apr 19 16:41:20 DEBUG[17683]: chan_sip.c:11840 sip_devicestate: Checking device state for peer myOpenSer
Apr 19 16:41:20 DEBUG[17683]: devicestate.c:187 do_state_change: Changing state for SIP/myOpenSer - state 2 (In use)
scp99*CLI>
Disconnected from Asterisk server

Btw:
Asterisk crashes also after a transfer (back) into PSTN.

Any help would be appreciated :smile:

Thanks kind regards,
Sancho

2

Additional info:
Same with fresh installed 1.2.17.

Kind regards,
Sancho

3

We had the same problem when using ExpressTalk softphone. Doing transfer or two on that would definetly crash our call center. Only solution was to use native asterisk transfer (configured in features.conf)

4

Well, it seems I am too silly to do the native transfer.

On the other hand: I built up an Asterisk in a call center environment and there transfers work perfectly with Grandstream GXP-2000s.

Well, anyway, I have new information:
After debugging and searching I found out that it seems that my PHP script has problems with the transfer.
The script is started with DeadAGI, checks some things and then does the dial. After the dial it does some more things (e.g. writing into a database) and finally terminates after sending a hangup to the Asterisk (Oh dear! While writing this, I fear it is the Hangup).
If I do not use any scripts, everything is fine.

So, I am on the way into the abyss of my scripts :wink:

Kind regards,
Sancho