- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 27 Mar '15

27 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

bind_meta_app - bind to * AND #
by Денис 27 Mar '15

27 Mar '15

Hello. I want use bind_meta_app, how i can bind on *2 and #1? If i set bind_meta_key=# - bind on 2 BEFORE this set - bound to #2. I don`t want use bind_digit_action - it's so strange and incomprehensible, especially bind to leg B My last try: <action application="bind_digit_action" data="att_xfer,*2,exec:execute_extension,att_xfer XML features"/> <action application="bind_meta_app" data="2 b b execute_extension::att_xfer XML features"/> <action application="set" data="bind_meta_key=#"/> <action application="bind_digit_action" data="blind_xfer,#1,exec:execute_extension,dx2 XML features"/> <action application="bind_meta_app" data="1 b b execute_extension::dx XML features"/> Who have working configs for transfer (blind+att) with A+B legs?

1 0

Swap on att_xfer application
by Jayesh Nambiar 27 Mar '15

27 Mar '15

Hello, I was exploring the usage of att_xfer and it works as expected. I was just wondering that is there a way when the last party answers, I can hold that party and get back to initial call placing the last party on hold using some DTMF action?? I see a DTMF action like # to end the last call and get back to the initial call, but what I am looking at is hold the last call and get back to initial call before I actually transfer the call. Is there anything similar available which can be used to swap between calls using the att_xfer application. Thanks, --- Jayesh

1 0

Call originate suggestion needed
by Andre DeMattia 27 Mar '15

27 Mar '15

HI, I need some suggestions on how to use Limit on an outbound gateway(s) and count the calls that make it past limit. Currently I have my limit fire on_originate and it works but it cause a Leg B cdr. (not a big deal) and I need to count the calls but when I count them on on_Post Originate only the successful ringing calls are counted, 503's etc are not counted. I can't count the calls on originate because that doesn't remove the exceeded limit calls So what I want to do is call my dialplan and before the calls start check limit and move on to the next bridge. Any call that makes it to the provider I need to count to keep track (cps, ports etc) Any suggestions? I don't want to do loopback since i'm trying to get the calls out of the switch as fast as possible. thanks all Andre

5 4

How do I bridge to a different profile AND gateway?
by Sebastian Auriol 26 Mar '15

26 Mar '15

Hi list, It's my first post: be gentle! ;-) I've read the wiki and I've searched the mailing list but I can't find out how to do what I want to do and that is: Bridge an incoming call to a different profile AND specify the gateway? I can easily see how to do one or the other but not both. Why would I want to do this? The incoming call is on the internal profile. The internal profile uses UDP and ACLs as it is all connected to a single switch. But the call from internal needs to be bridged to an external gateway using TLS - on a different network interface. So it should be on a different profile. However, I don't want to specify the IP address in the dialplan because the destination is another gateway, so it should be a gateway. Any help appreciated - I'm fairly new to FS. Kind regards, SebA

3 2

possible bug?
by David Villasmil Govea 26 Mar '15

26 Mar '15

Hello guys, I had this situation in which I'm generating calls to FS from port 5061, but on the actual INVITE it says port 5060. FS is responding to port 5060 instead of port 5061 which is the port from which it received the requests. Is this by design? Regards, David -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Help Mod_Nibblebill
by Antony Aguirre Morales 26 Mar '15

26 Mar '15

I have a problem with my configuration of nibblebill module, is configured so that when the cash = 0 send the call to hangup, but brand sip Error 483 - Too many hops. My configuration is: mod_nibblebill: <param name = "nobal_amt" value = "0" /> <param name = "nobal_action" value = "hangup XML default" /> I have created in the dialplan extencion the following: <extension name = "hangup"> <condition field = "destination_number" expression = "^ (hangup) $"> <action application = "playback" data = "voicemail / vm-continue.wav" /> <action application = "hangup" /> </ condition> </ extension> but does not work in my error log I find the following: Dialplan: sofia/internal/8117(a)1.1.1.1 Regex (PASS) [mierda] destination_number(mierda) =~ /^(mierda)$/ break=on-false Dialplan: sofia/internal/8117(a)1.1.1.1 Action playback(voicemail/vm-continue.wav) Dialplan: sofia/internal/8117(a)1.1.1.1 Action hangup() 2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/8117(a)1.1.1.1) State Change CS_ROUTING -> CS_EXECUTE 2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/8117(a)1.1.1.1[BREAK] 2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/8117(a)1.1.1.1) State ROUTING going to sleep 2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/8117(a)1.1.1.1) Running State Change CS_EXECUTE 2015-03-25 12:16:38.365165 [DEBUG] switch_channel.c:2169 (sofia/internal/ 8117(a)1.1.1.1) Callstate Change RINGING -> EARLY 2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/8117(a)1.1.1.1) State EXECUTE 2015-03-25 12:16:38.365165 [DEBUG] mod_sofia.c:243 sofia/internal/ 8117(a)1.1.1.1 SOFIA EXECUTE 2015-03-25 12:16:38.365165 [INFO] switch_core_session.c:1511 sofia/internal/ 8117(a)1.1.1.1 setting session heartbeat to 60 second(s). 2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:209 sofia/internal/8117(a)1.1.1.1 Standard EXECUTE EXECUTE sofia/internal/8117(a)1.1.1.1 set(open=true) 2015-03-25 12:16:38.365165 [DEBUG] mod_dptools.c:1402 sofia/internal/ 8117(a)1.1.1.1 SET [open]=[true] EXECUTE sofia/internal/8117(a)1.1.1.1 playback(voicemail/vm-continue.wav) 2015-03-25 12:16:38.365165 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:643 Received request via SESSION_HEARTBEAT! 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:486 Attempting to bill at $1 per minute to account 1 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:498 Not billing 1 - call is not in answered state 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:418 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:426 Retrieved current balance for account 1 (balance = 0.000000) 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000 to hangup balance of 0.000000 2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000 fell below allowed amount of 0.000000! (Account 1) 2015-03-25 12:16:38.365165 [DEBUG] switch_ivr.c:1834 (sofia/internal/ 8117(a)1.1.1.1) State Change CS_EXECUTE -> CS_ROUTING 2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/8117(a)1.1.1.1[BREAK] 2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/8117(a)1.1.1.1[BREAK] 2015-03-25 12:16:38.365165 [NOTICE] switch_ivr.c:1841 Transfer sofia/internal/8117(a)1.1.1.1 to XML[mierda@default] 2015-03-25 12:16:38.385162 [DEBUG] switch_ivr_play_say.c:1708 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-continue.wav 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/8117(a)1.1.1.1) State EXECUTE going to sleep 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/8117(a)1.1.1.1) Running State Change CS_ROUTING 2015-03-25 12:16:38.385162 [DEBUG] switch_channel.c:2165 (sofia/internal/ 8117(a)1.1.1.1) Callstate Change EARLY -> RINGING 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/8117(a)1.1.1.1) State ROUTING 2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:150 sofia/internal/ 8117(a)1.1.1.1 SOFIA ROUTING 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:486 Attempting to bill at $1 per minute to account 1 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:498 Not billing 1 - call is not in answered state 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current balance for account 1 (balance = 0.000000) 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000 to hangup balance of 0.000000 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000 fell below allowed amount of 0.000000! (Account 1) 2015-03-25 12:16:38.385162 [NOTICE] switch_ivr.c:1736 Hangup sofia/internal/ 8117(a)1.1.1.1 [CS_ROUTING] [EXCHANGE_ROUTING_ERROR] 2015-03-25 12:16:38.385162 [DEBUG] switch_channel.c:3189 Send signal sofia/internal/8117(a)1.1.1.1 [KILL] 2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/8117(a)1.1.1.1[BREAK] 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current balance for account 1 (balance = 0.000000) 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/8117(a)1.1.1.1) State ROUTING going to sleep 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/8117(a)1.1.1.1) Running State Change CS_HANGUP 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:681 (sofia/internal/8117(a)1.1.1.1) Callstate Change RINGING -> HANGUP 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:683 (sofia/internal/8117(a)1.1.1.1) State HANGUP 2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:506 Channel sofia/internal/ 8117(a)1.1.1.1 hanging up, cause: EXCHANGE_ROUTING_ERROR 2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:640 Responding to INVITE with: 483 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:486 Attempting to bill at $1 per minute to account 1 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:498 Not billing 1 - call is not in answered state 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current balance for account 1 (balance = 0.000000) 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000 to hangup balance of 0.000000 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000 fell below allowed amount of 0.000000! (Account 1) 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1'] 2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current balance for account 1 (balance = 0.000000) 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:48 sofia/internal/8117(a)1.1.1.1 Standard HANGUP, cause: EXCHANGE_ROUTING_ERROR 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:683 (sofia/internal/8117(a)1.1.1.1) State HANGUP going to sleep 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:450 (sofia/internal/8117(a)1.1.1.1) State Change CS_HANGUP -> CS_REPORTING 2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/8117(a)1.1.1.1[BREAK] 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/8117(a)1.1.1.1) Running State Change CS_REPORTING 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:767 (sofia/internal/8117(a)1.1.1.1) State REPORTING 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:92 sofia/internal/8117(a)1.1.1.1 Standard REPORTING, cause: EXCHANGE_ROUTING_ERROR 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:767 (sofia/internal/8117(a)1.1.1.1) State REPORTING going to sleep 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:444 (sofia/internal/8117(a)1.1.1.1) State Change CS_REPORTING -> CS_DESTROY 2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/8117(a)1.1.1.1[BREAK] 2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1559 Session 201 (sofia/internal/8117(a)1.1.1.1) Locked, Waiting on external entities 2015-03-25 12:16:38.385162 [NOTICE] switch_core_session.c:1577 Session 201 (sofia/internal/8117(a)1.1.1.1) Ended 2015-03-25 12:16:38.385162 [NOTICE] switch_core_session.c:1581 Close Channel sofia/internal/8117(a)1.1.1.1[CS_DESTROY] 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:572 (sofia/internal/8117(a)1.1.1.1) Running State Change CS_DESTROY 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:582 (sofia/internal/8117(a)1.1.1.1) State DESTROY 2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:399 sofia/internal/ 8117(a)1.1.1.1 SOFIA DESTROY 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:99 sofia/internal/8117(a)1.1.1.1 Standard DESTROY 2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:582 (sofia/internal/8117(a)1.1.1.1) State DESTROY going to sleep any ideas? The design is based on a cost for all of the following linkhttps:// freeswitch.org/confluence/display/FREESWITCH/mod_nibblebill#mod_nibblebill-…

1 0

Support for actual Silence Suppression
by Markus von Arx 26 Mar '15

26 Mar '15

Hi I'd like to configure FreeSWITCH in such a way that it does actual real silence suppression and reduce the used network bandwidth during silence periods. By actual real silence suppression I mean that it uses one of the two options described in RFC 3389 chapter 5, so either 1) Sending RTP packets with payload 13 that carry CNG description instead of PCMU RTP packets during silence periods, and send those RTP payload 13 packets much less often than every 20 milliseconds, or 2) Use discontinuous RTP transmission during silence periods, or in other words send PCMU RTP packets much less often than every 20 milliseconds. So my questions: 1) Does FreeSWITCH support on of those two options described above? 2) If yes, how do I configure FreeSWITCH to actually do it? Btw: I do know about https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG, and I have tried most of the things described there. What I found (by using Wireshark) is that FreeSWITCH does indeed support VAD and CNG, but that the FreeSWITCH does not use that information to send RTP payload 13 CNG packets nor to do discontinuous RTP transmission, but instead the FreeSWITCH continues to send PCMU RTP packets every 20 milliseconds and puts the CNG audio inside these PCMU packets. This "works" in the way that the remote party hears CNG, but this is not silence suppression as it does not reduce the network bandwidth. Thanks a lot, Markus

1 0

Silence Suppression from an Audio Conference
by Markus von Arx 26 Mar '15

26 Mar '15

Hi Can anyone tell me if FreeSWITCH supports silence suppression for SIP calls that are inside a FreeSWITCH audio conference? If yes, how do I configure mod_conference, mod_sofia and FreeSWITCH core to enable this feature? More precisely, I try to enable/activate the behavior described in RFC 3389 for G.711 in such a way that there are only RTP packets of type 13 every 1 or 2 seconds. I tried to play around with some possible settings but could never observe anything else then the regular G.711 PCMU RTP packets on the wire. Even when I set the SIP call to 'deaf' via the FreeSWITCH console, mod_conference/mod_sofia continue to send G.711 PCMU RTP packets every 20ms. It's possible that I completly misunderstand RFC 3389 and the concepts of silence suppression, comfort noise etc. In the end, what I try to achieve is to reduce the network bandwidth of a G.711 SIP channel during the periods when the FreeSWITCH only sends silence over the SIP channel. Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to another codec. Thanks, Markus

3 5

eliminating fax header line
by Bruce Lefko 26 Mar '15

26 Mar '15

I'd like to completely eliminate the header line sent at the top of each fax page using spandsp. i've set fax-header to a null string, but that doesn't seem to be removing it, and it defaults back to 'Spandsp fax header'. This seems to be the default in the source code in 1.2. What do I have to do to eliminate the header line? Do I have to make changes to freeswitch source? Thanks!

6 7