June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ReINVITE failure with ACK+SDP scenario
by BNML 02 Jul '13

02 Jul '13

Hi all, I am having trouble with a transfert failure in a special scenario which i wasn't able to find documentation or configuration option solving it. I have a PBX creating two outbound call, it want to bridge them on the FreeSWITCH. It is sending a first INVITE without SDP and it will send the SDP in the ACK. (This is not breaking RFC3261 but I admit rarely seen scenario. http://tools.ietf.org/html/rfc3261#section-13.2.1 : "The UAC MAY choose to add a message body to the INVITE.". Looking at this scenario there is, indeed, at least one interesting scenario where it can helps avoiding creating another useless Re-INVITE transaction, see below) For the second call it is sending a re-invite with SDP. FreeSWITCH will understand and forward (because of "bypass_media") the second Re-INVITE but does never forward the first Re-INVITE (The one without SDP in the INVITE but in the ACK) to carrier, so this will end up in one-way-audio. A little schema to help : PBX FREESWITCH COMMENT 1) INVITE (a-leg) [THERE IS NO SDP] 2) 200 OK (fs-a-leg)[FreeSWITCH answer with the SDP information before any media update occurs, but my PBX is now aware of media informations] 3) INVITE+SDP (b-leg) [This is b-leg updated media information with previous 200 OK (fs-a-leg)] 4) 200 OK (fs-b-leg)[My PBX updated media and has b-leg freeswitch media information that it will send in the ACK+SDP to finalize the both-legs Re-INVITE] 5) ACK (b-leg) [Acknoledgement of b-leg Re-INVITE transaction] 6) ACK+SDP (a-leg) [ *** problem *** it assume FreeSWITCH will update media information and forward to the carrier which it won't] The solution i imagine is FreeSWITCH SHOULD trigger a Re-INVITE either for the a-leg at INVITE (without SDP in "1") receiption (and update media information when receiving the ACK+SDP) _OR_ create a Re-INVITE transaction at "6" when receiving ACK+SDP. One way or another, I am supposing FreeSWITCH refuse to forward the INVITE without SDP at first maybe because of carrier compatibility ? If not, this seems the way to go. Has anyone faced such a scenario ? Am I rigth assuming a patch is needed ? If yes, any suggestions are welcome. Thanks you BNML

3 6

Updated 1.4.beta tarball
by Ken Rice 02 Jul '13

02 Jul '13

Hey guys, Check it out http://files.freeswitch.org/freeswitch-1.4.0.beta4.tar.bz2 Its out on the CDN... If you are tracking the v1.4.beta branch in git, and have updated recently you¹re on track! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

5 8

What am I looking for?
by Karl Schmidt 01 Jul '13

01 Jul '13

I am looking to migrate out of an old KX-T61610 system. I plan to support up to 16 old analog phone lines - so I'm looking for something I can plug into my server provides 16 analog extension connections. ( Best would be something that supported the old Panasonic KX-t phones - but I doubt that exists.) -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Gumption is 99% of success. kps --------------------------------------------------------------------------------

4 4

What is rtp-autofix-timing
by Mino Haluz 01 Jul '13

01 Jul '13

Hi, can somebody explain me what parameter rtp-autofix-timing does in fact? I had to set this parameter to false in order to avoid warnings with asynchronous ptime not supported error, but I cant get any information about this on wiki. Thanks, M

2 1

call_timeout and lua in dialplan
by Vincent Xia 01 Jul '13

01 Jul '13

in a dialplan extension, i have <action application="set" data="call_timeout=30" /> then <action application="lua" data="bridge.lua ${effective_caller_id_number} $1"> in which call bridging is to be processed, after this, is <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}" /> to transfer the call to voicemail. the purpose is to test no-answer till timeout and see if the call is transferred as planned. but nothing happened untill 60 secs after the dial, which is the default value of call_timeout. in my understanding, when lua takes over, the channel variable call_timeout is no longer effective and will not affect what happens in lua script, so i tried session:execute("set", "call_timeout=30"); in bridge.lua, but the problem is still there. how to make call_timeout work with lua, could anyone figure this out?

2 2

music on hold problem with Sangoma A200 board on FS
by Tamer Higazi 30 Jun '13

30 Jun '13

Hi people! I am running FreeSWITCH Version 1.5.2b+git~20130618T211908Z~9cf05e3595 (git 9cf05e3 2013-06-18 21:19:08Z) and I have a problem with music on hold with my Sangoma A200 board, and I don't know how to solve it. Everytime, the other party puts me on hold, I don't hear anything and the connection is dropped. Any ideas?! For any ideas solving this problem, I woudl kindly thank you. Tamer My freetdm.conf.xml looks like this: configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${hold_music}"/> <param name="enable-analog-option" value="call-swap"/> <param name="enable-analog-option" value="3-way"/> </settings> <config_profiles> </config_profiles> <analog_spans> <span name="FXS">  <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> </analog_spans> </configuration> and the console output is here: 2013-06-27 16:44:13.286747 [DEBUG] switch_channel.c:1746 (sofia/external/061312000) Callstate Change ACTIVE -> HELD 2013-06-27 16:44:13.286747 [DEBUG] switch_core_session.c:1141 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:13.286747 [DEBUG] switch_core_session.c:861 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:13.336605 [DEBUG] switch_ivr.c:612 FreeTDM/1:1/0061312000 Command Execute playback(local_stream://moh) EXECUTE FreeTDM/1:1/0061312000 playback(local_stream://moh) 2013-06-27 16:44:13.336605 [WARNING] mod_local_stream.c:469 Unknown source moh, trying 'default' 2013-06-27 16:44:13.336605 [ERR] mod_local_stream.c:478 Unknown source default 2013-06-27 16:44:13.336605 [DEBUG] switch_core_session.c:861 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:13.336605 [DEBUG] switch_ivr_bridge.c:388 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:2723 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:10:64000] 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:1748 Already using PCMA 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:2885 Set 2833 dtmf send payload to 101 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:3885 Audio params are unchanged for sofia/external/061312000. 2013-06-27 16:44:13.536666 [DEBUG] switch_core_media.c:3895 sofia/external/061312000 Setting audio receive payload in Re-INVITE to 8 2013-06-27 16:44:13.536666 [DEBUG] sofia.c:6286 Processing updated SDP 2013-06-27 16:44:13.536666 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.546766 [DEBUG] sofia.c:5745 Channel sofia/external/061312000 entering state [completed][200] 2013-06-27 16:44:13.606614 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.606614 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.606614 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:13.616646 [DEBUG] sofia.c:5745 Channel sofia/external/061312000 entering state [ready][200] 2013-06-27 16:44:23.186805 [DEBUG] ftmod_wanpipe.c:1678 [s1c1][1:3] read wanpipe event 5 2013-06-27 16:44:23.186805 [DEBUG] ftmod_wanpipe.c:1462 [s1c1][1:3] Got wanpipe ONHOOK 2013-06-27 16:44:23.186805 [DEBUG] ftmod_wanpipe.c:1685 [s1c1][1:3] Ignoring event for now 2013-06-27 16:44:23.936625 [DEBUG] ftmod_wanpipe.c:1661 [s1c1][1:3] Diff since last event = 752 ms, delivering ONHOOK now 2013-06-27 16:44:23.936625 [DEBUG] ftmod_analog.c:999 [s1c1][1:3] Received event [ONHOOK] in state [UP] 2013-06-27 16:44:23.936625 [DEBUG] ftmod_analog.c:1052 [s1c1][1:3] Changed state from UP to DOWN 2013-06-27 16:44:23.946747 [DEBUG] ftmod_analog.c:638 [s1c1][1:3] Completed state change from UP to DOWN in 6 ms 2013-06-27 16:44:23.946747 [DEBUG] ftmod_analog.c:644 [s1c1][1:3] Executing state handler on 1:1 for DOWN 2013-06-27 16:44:23.946747 [DEBUG] mod_freetdm.c:2322 got FXS sig [STOP] 2013-06-27 16:44:23.946747 [NOTICE] mod_freetdm.c:2414 Hangup FreeTDM/1:1/0061312000 [CS_EXECUTE] [NORMAL_CLEARING] 2013-06-27 16:44:23.946747 [DEBUG] switch_channel.c:3135 Send signal FreeTDM/1:1/0061312000 [KILL] 2013-06-27 16:44:23.946747 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:3180 [s1c1][1:3] DTMF debug is already disabled 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:3212 [s1c1][1:3] No need to disable input dump 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:3243 [s1c1][1:3] No need to disable output dump 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:6798 Cleared call with id 12 2013-06-27 16:44:23.946747 [DEBUG] ftdm_io.c:2985 [s1c1][1:3] channel done 2013-06-27 16:44:23.946747 [DEBUG] ftmod_analog.c:972 [s1c1][1:3] ANALOG CHANNEL 1:1 thread ended. 2013-06-27 16:44:23.946747 [WARNING] ftdm_io.c:4286 [s1c1][1:3] cannot read from channel that is not open 2013-06-27 16:44:23.946747 [WARNING] mod_freetdm.c:792 Failed to read from channel FreeTDM/1:1/0061312000 device 1:1! 2013-06-27 16:44:23.946747 [DEBUG] switch_ivr_bridge.c:642 BRIDGE THREAD DONE [FreeTDM/1:1/0061312000] 2013-06-27 16:44:23.946747 [DEBUG] switch_ivr_bridge.c:667 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_ivr_bridge.c:642 BRIDGE THREAD DONE [sofia/external/061312000] 2013-06-27 16:44:23.956746 [DEBUG] switch_ivr_bridge.c:667 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.956746 [NOTICE] switch_ivr_bridge.c:730 Hangup sofia/external/061312000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-06-27 16:44:23.956746 [DEBUG] switch_channel.c:3135 Send signal sofia/external/061312000 [KILL] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:481 (sofia/external/061312000) State EXCHANGE_MEDIA going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (sofia/external/061312000) Running State Change CS_HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (sofia/external/061312000) State HANGUP 2013-06-27 16:44:23.956746 [DEBUG] mod_sofia.c:463 Channel sofia/external/061312000 hanging up, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] mod_sofia.c:515 Sending BYE to sofia/external/061312000 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:48 sofia/external/061312000 Standard HANGUP, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (sofia/external/061312000) State HANGUP going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:692 (sofia/external/061312000) Callstate Change HELD -> HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:447 (sofia/external/061312000) State Change CS_HANGUP -> CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (sofia/external/061312000) Running State Change CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (sofia/external/061312000) State REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:92 sofia/external/061312000 Standard REPORTING, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (sofia/external/061312000) State REPORTING going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:441 (sofia/external/061312000) State Change CS_REPORTING -> CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1549 Session 24 (sofia/external/061312000) Locked, Waiting on external entities 2013-06-27 16:44:23.956746 [DEBUG] switch_ivr_bridge.c:1531 FreeTDM/1:1/0061312000 skip receive message [UNBRIDGE] (channel is hungup already) 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1567 Session 24 (sofia/external/061312000) Ended 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/061312000 [CS_DESTROY] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:2739 FreeTDM/1:1/0061312000 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:478 (FreeTDM/1:1/0061312000) State EXECUTE going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/1:1/0061312000) Running State Change CS_HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/1:1/0061312000) State HANGUP 2013-06-27 16:44:23.956746 [DEBUG] mod_freetdm.c:544 [1:1] FreeTDM/1:1/0061312000 CHANNEL HANGUP ENTER 2013-06-27 16:44:23.956746 [DEBUG] mod_freetdm.c:569 Device [1:1] is no longer attached to FreeTDM/1:1/0061312000. Nothing to do. 2013-06-27 16:44:23.956746 [DEBUG] mod_freetdm.c:640 [1:1] FreeTDM/1:1/0061312000 CHANNEL HANGUP EXIT 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:48 FreeTDM/1:1/0061312000 Standard HANGUP, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/1:1/0061312000) State HANGUP going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/1:1/0061312000) Callstate Change ACTIVE -> HANGUP 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:566 (sofia/external/061312000) Callstate Change HANGUP -> DOWN 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/1:1/0061312000) State Change CS_HANGUP -> CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/1:1/0061312000) Running State Change CS_REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/1:1/0061312000) State REPORTING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:569 (sofia/external/061312000) Running State Change CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (sofia/external/061312000) State DESTROY 2013-06-27 16:44:23.956746 [DEBUG] mod_sofia.c:373 sofia/external/061312000 SOFIA DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:99 sofia/external/061312000 Standard DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (sofia/external/061312000) State DESTROY going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:92 FreeTDM/1:1/0061312000 Standard REPORTING, cause: NORMAL_CLEARING 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/1:1/0061312000) State REPORTING going to sleep 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/1:1/0061312000) State Change CS_REPORTING -> CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/1:1/0061312000 [BREAK] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_session.c:1549 Session 23 (FreeTDM/1:1/0061312000) Locked, Waiting on external entities 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1567 Session 23 (FreeTDM/1:1/0061312000) Ended 2013-06-27 16:44:23.956746 [NOTICE] switch_core_session.c:1571 Close Channel FreeTDM/1:1/0061312000 [CS_DESTROY] 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:566 (FreeTDM/1:1/0061312000) Callstate Change HANGUP -> DOWN 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:569 (FreeTDM/1:1/0061312000) Running State Change CS_DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (FreeTDM/1:1/0061312000) State DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:99 FreeTDM/1:1/0061312000 Standard DESTROY 2013-06-27 16:44:23.956746 [DEBUG] switch_core_state_machine.c:579 (FreeTDM/1:1/0061312000) State DESTROY going to sleep freeswitch@internal>

1 0

Return raw file.
by Mimiko 30 Jun '13

30 Jun '13

Hello. I want to use mod_xml_rpc, or something else to return raw audio files which a recorded on file system using lua script to verify credentials and select file. For now I enable mod_xml_rpc and created a simple lua script: local f=assert(io.open("/path/file.mp3","rb")) io.write( f:read("*all")) f:close() Using http://freeswitch:ip:8080/api/luarun?script.lua gives "+OK", but not the file content. Using http://freeswitch:ip:8080/api/lua?script.lua gives error. Instead I change: local f=assert(io.open("/path/file.mp3","rb")) stream:write( f:read("*all")) f:close() gives nothing thru mod_xml_rpc, but on console outputs the content of the file if file is text. If file is mp3, the output is just some rando three characters. How to get the whole content fro an audio file using lua and some access method? -- Mimiko desu.

2 2

unknown event 22: 988 Incomplete offer/answer
by Mino Haluz 30 Jun '13

30 Jun '13

Hi, I have two 2 FS connected this way: caller(g729) - FS1 - FS2 - calee(g711) Everything works fine, unless I set proxy_media on FS2. And I want to set it because I do not want to transcode twice (g729-linear-g729 and g729-linear-g711). So calee rings, sends 200OK to FS2 with correct SDP v=0 o=- 197090 197090 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=audio 16392 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv m=audio 0 RTP/AVP 4 101 13 and FS2 changes SDP this way and sends to FS1 v=0 o=- 157558 157558 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=audio 16386 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=audio 0 RTP/AVP 4 96 13 a=rtpmap:96 /0 FS1, writes to log: sofia.c:1468 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer and sends bye to callee and 500 No session set by user to caller. What should I do in this case? Is there any problem with SDP format? Thanks, M

1 0

modfifo not calling all registered members
by efern75＠hushmail.com 30 Jun '13

30 Jun '13

Good evening all, Have an issue with modfifo not calling all members. This has been happening for a while but can't find out why. I have 10 yealink phones/1 snom phone that have each 3 individual registered extension. (one phone has ext: 101,201,301, the next phone has 102,202,302, etc). I have all 33 extensions registered according to "sofia status profile internal reg". I add each extension to fifo via a script that just runs: fifo_member add FIFO0 {fifo_member_wait=nowait}user/EXTENSION Below is the pastebin of "fifo list" after loading the above command against all registered phones (33 extensions,11 phones) http://pastebin.com/diNU6hbm This is how the fifo.conf.xml file looks like (default): <configuration name="fifo.conf" description="FIFO Configuration"> <settings> <param name="delete-all-outbound-member-on-startup" value="false"/> </settings> <fifos> </fifos> </configuration> When a call comes in and is sent to the FIFO queue via: <action application="fifo" data="FIFO0 in"/> Only 23 channels/call setups are made to the possible 33 extensions that are in fifo list. below is the show channels of this setup: http://pastebin.com/s7TvxGMa Here is a list of the 10 non called extension: 102,305,205,123,308,201,101,206,306,222 After hanging up, this is the "fifo list" dump for reference: http://pastebin.com/9JgD49fL I have ran "fifo debug 7" and /debug 7 at the console and did some test calls and nowhere does it light up with the extensions that are missing for the fifo call setup =\ I have also for example ran ngrep against a phone that only has two out of three INVITES sent (based on show channels) and can verify that only 2 INVITES are being sent. (just wanted to make sure that the phone wasn't rejecting the invite on the missing show channels extension due not not being very fast at seeing it happen). I am running: OS: Debian 6.0.7 (x64) FS: 1.2 latest git (1.2.10+git~20130628T192118Z~929f52b15c) Also, i have shutdown freeswitch, deleted the entire /usr/local/freeswitch/db/ directory content; started freeswitch, waited for all extensions to fill in eventually, and ran the fifo_member add command and can still duplicate this issue =\ Thank you for your assistance on this!!!! Eric

1 0

Limit usage not decreased after attended transfer
by Stanislav Sinyagin 29 Jun '13

29 Jun '13

hi, My user wants to have only one incoming call on a number of his SIP accounts, and no concurrent calls. So, on incoming calls I do <action application="limit" data="hash ${domain} ${availability_username} 1"/> and on outgoing calls I do <action application="limit" data="hash ${domain_name} ${availability_username} -1"/> Problem: when an inbound call arrives, and the user transfers it to another PSTN number and hangs up, limit_usage still shows 1 active conversation, until the transferred call ends. That means, the user already hung up, but he cannot accept new calls until the original call finishes. Any idea how to eliminate this situation and free up the limit counts after transfer? thanks stanislav

1 1