April 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

4 8

Limit Database
by Andre 05 Aug '14

05 Aug '14

HI, on Windows I'm trying to use Limit with Microsoft SQL DB but it's still using the sqlight database. I have mod_Db and Mod_Hash loaded and under db.config.xml I have my odbc setup like this <param name="odbc-dsn" value="sqldb64::" /> I know the odbc works because I have the core database setup the same <param name="odbc-dsn" value="sqldb64::" /> <param name="core-dbtype" value="MSSQL"/> Is there a step that I'm missing? Thanks

11 20

Flowroute is testing CNAM lookup
by Steven Schoch 27 May '14

27 May '14

I am one of the Beta testers of Flowroute's CNAM lookup for inbound calls. So far, it seems to be working quite well. -- Steve

3 4

Voicemails chopped at 10 seconds (randomly!)
by Peter Hartmann 27 May '14

27 May '14

Hey, I have a user that randomly gets voicemails chopped off at exactly 10 seconds. This is using a Sangoma A200 analog card....so it's not a NAT problem. Apparently this has nothing to do with silence detection as I've tried different settings for that. Cutoffs happen in mid word and the audio file length is exactly the same in each when it happens....not always. There's nothing unique about the way and which calls get to this users mailbox. IVR does this: "transfer 1201 XML default". Users mailbox is 1201 and the caller is redacted to 212xxxxxxx. Here's the FS Log for one of the calls that had a 10 second voicemail. http://pastebin.com/Fq5N9UCq Here's Local Extension dialplan snippet that the calls pass through in the event I'm doing something wrong there. http://pastebin.com/FK0zXs4A Admittedly this is a bit out of date: FreeSWITCH Version 1.5.8b+git~20140124T004436Z~bd7182c390~64bit (git bd7182c 2014-01-24 00:44:36Z 64bit) I'm perplexed! Thank very much in advance! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it.

3 4

Debian Shell Script for Installing FreeSWITCH
by Joel White 15 May '14

15 May '14

https://gist.github.com/zhacker98/9643809 I would suggest running under 'screen' as the process takes a moment

5 16

Windows Users
by Brian West 15 May '14

15 May '14

Can windows users help verify and update if things are out of date? Mario transitioned it but I want the windows users to help verify these. https://confluence.freeswitch.org/display/FREESWITCH/Windows Thanks, -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

2 2

wrong q.850
by Vladimir Getmanshchuk 14 May '14

14 May '14

Hello Everyone! I have faced same issue as Ross at this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/066985.… FS translates q.850 codes from b-leg to a-leg incorrect. Looks like there is no solution to translate q850 in correct way? Please advice! Additional info: # freeswitch -version FreeSWITCH version: 1.2.11+git~20130711T181514Z~2eab94be78 (git 2eab94b 2013-07-11 18:15:14Z) dialplan extension: <extension name="default"> <condition field="destination_number" expression="^\+?222222222222"> <action application="bridge" data="{max_forwards=70,progress_timeout=30,hangup_after_bridge=true,sip_cid_type=pid,zrtp_secure_media=false,hold_music=silence[absolute_codec_string=^^:G729@20i :PCMA@20i:PCMU@20i ,origination_caller_id_number=+11111111111,origination_caller_id_name=+111111111111]sofia/external/ 222222222222@2.2.2.2:5062"/> </condition> </extension> SIP trace: # U 2014/04/10 10:19:43.079198 2.2.2.2:5062 -> 1.1.1.1:5080 SIP/2.0 480 Temporarily not available. Via: SIP/2.0/UDP 1.1.1.1:5080;rport=5080;branch=z9hG4bKBXQm2F3cHgFcK. From: "+111111111111" <sip:+111111111111@1.1.1.1>;tag=t8yp14HU53Xrj. To: <sip:+222222222222@4.4.4.4:5060>;tag=gK0699dffd. Call-ID: 5f60343b-3b3c-1232-d9ba-020820d8e22d. CSeq: 58226505 INVITE. Reason: Q.850;cause=19. Content-Length: 0. . # U 2014/04/10 10:19:43.084570 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK26dc.1612d515.0. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK420c53ac;rport=5060. Max-Forwards: 16. From: "pbx234" <sip:111111111111@3.3.3.3>;tag=as333cf884. To: <sip:0599#222222222222@2.2.2.2:5060>;tag=veXaXcyc8118e. Call-ID: 6c86be36267a0f6d54a34060364a3490@3.3.3.3:5060. CSeq: 102 INVITE. User-Agent: FS. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "Outbound Call" <sip:222222222222@2.2.2.2 >;party=calling;privacy=off;screen=no. *2014-04-10 10:50:03.470035 [DEBUG] sofia.c:4937 Remote Reason: 192014-04-10 10:50:03.490051 [ALERT] switch_ivr.c:828 sofia/external/222222222222@2.2.2.2:5062 <http://222222222222@2.2.2.2:5062> receive message [SIGNAL_DATA] 2014-04-10 10:50:03.490051 [DEBUG] sofia.c:5719 Channel sofia/external/222222222222@2.2.2.2:5062 <http://222222222222@2.2.2.2:5062> entering state [terminated][480]2014-04-10 10:50:03.490051 [NOTICE] sofia.c:6528 Hangup sofia/external/222222222222@2.2.2.2:5062 <http://222222222222@2.2.2.2:5062> [CS_EXCHANGE_MEDIA] [NO_ANSWER]* 2014-04-10 10:50:03.490051 [DEBUG] switch_channel.c:3130 Send signal sofia/external/222222222222@2.2.2.2:5062 [KILL] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_session.c:1334 Send signal sofia/external/222222222222@2.2.2.2:5062 [BREAK] 2014-04-10 10:50:03.490051 [ALERT] switch_ivr.c:828 sofia/external/ 222222222222@2.2.2.2:5062 receive message [SIGNAL_DATA] 2014-04-10 10:50:03.490051 [DEBUG] switch_ivr_bridge.c:557 sofia/external/ 222222222222@2.2.2.2:5062 ending bridge by request from write function 2014-04-10 10:50:03.490051 [DEBUG] switch_ivr_bridge.c:633 BRIDGE THREAD DONE [sofia/external/222222222222@2.2.2.2:5062] 2014-04-10 10:50:03.490051 [DEBUG] switch_ivr_bridge.c:658 Send signal sofia/internal/111111111111@1.1.1.1:5060 [BREAK] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:480 (sofia/external/222222222222@2.2.2.2:5062) State EXCHANGE_MEDIA going to sleep 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:415 (sofia/external/222222222222@2.2.2.2:5062) Running State Change CS_HANGUP 2014-04-10 10:50:03.490051 [DEBUG] switch_ivr_bridge.c:633 BRIDGE THREAD DONE [sofia/internal/111111111111@1.1.1.1:5060] 2014-04-10 10:50:03.490051 [DEBUG] switch_ivr_bridge.c:658 Send signal sofia/external/222222222222@2.2.2.2:5062 [BREAK] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:678 (sofia/external/222222222222@2.2.2.2:5062) State HANGUP 2014-04-10 10:50:03.490051 [DEBUG] mod_sofia.c:504 Channel sofia/external/ 222222222222@2.2.2.2:5062 hanging up, cause: NO_ANSWER 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:48 sofia/external/222222222222@2.2.2.2:5062 Standard HANGUP, cause: NO_ANSWER 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:678 (sofia/external/222222222222@2.2.2.2:5062) State HANGUP going to sleep 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:689 (sofia/external/222222222222@2.2.2.2:5062) Callstate Change EARLY -> HANGUP 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:446 (sofia/external/222222222222@2.2.2.2:5062) State Change CS_HANGUP -> CS_REPORTING 2014-04-10 10:50:03.490051 [DEBUG] switch_core_session.c:1334 Send signal sofia/external/222222222222@2.2.2.2:5062 [BREAK] 2014-04-10 10:50:03.490051 [ALERT] switch_ivr_bridge.c:1521 sofia/external/ 222222222222@2.2.2.2:5062 receive message [UNBRIDGE] 2014-04-10 10:50:03.490051 [DEBUG] switch_ivr_bridge.c:1521 sofia/external/ 222222222222@2.2.2.2:5062 skip receive message [UNBRIDGE] (channel is hungup already) 2014-04-10 10:50:03.490051 [ALERT] switch_ivr_bridge.c:1524 sofia/internal/ 111111111111@1.1.1.1:5060 receive message [UNBRIDGE] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_session.c:854 Send signal sofia/internal/111111111111@1.1.1.1:5060 [BREAK] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:415 (sofia/external/222222222222@2.2.2.2:5062) Running State Change CS_REPORTING 2014-04-10 10:50:03.490051 [ALERT] switch_core_session.c:2732 sofia/internal/111111111111@1.1.1.1:5060 receive message [APPLICATION_EXEC_COMPLETE] 2014-04-10 10:50:03.490051 [NOTICE] switch_core_state_machine.c:262 sofia/internal/111111111111@1.1.1.1:5060 has executed the last dialplan instruction, hanging up. 2014-04-10 10:50:03.490051 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/internal/111111111111@1.1.1.1:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:761 (sofia/external/222222222222@2.2.2.2:5062) State REPORTING 2014-04-10 10:50:03.490051 [DEBUG] switch_channel.c:3130 Send signal sofia/internal/111111111111@1.1.1.1:5060 [KILL] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_session.c:1334 Send signal sofia/internal/111111111111@1.1.1.1:5060 [BREAK] 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/111111111111@1.1.1.1:5060) State EXECUTE going to sleep 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/111111111111@1.1.1.1:5060) Running State Change CS_HANGUP 2014-04-10 10:50:03.490051 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/111111111111@1.1.1.1:5060) State HANGUP 2014-04-10 10:50:03.490051 [DEBUG] mod_sofia.c:498 sofia/internal/ 111111111111@1.1.1.1:5060 Overriding SIP cause 480 with 480 from the other leg *2014-04-10 10:50:03.490051 [DEBUG] mod_sofia.c:504 Channel sofia/internal/111111111111@1.1.1.1:5060 <http://111111111111@1.1.1.1:5060> hanging up, cause: NORMAL_CLEARING2014-04-10 10:50:03.490051 [DEBUG] mod_sofia.c:638 Responding to INVITE with: 480* -- Yours sincerely, Vladimir Getmanshchuk

4 4

JIRA Reminder
by Brian West 06 May '14

06 May '14

FreeSWITCHers, When opening a JIRA and it asks for the Master git rev, it means one from the past 6 hours, In addition when opening a JIRA please be responsive. ‘Yes’ isn’t a valid answer, neither is a git rev from 30 days ago. Remember we are working to help you so at the very least do the leg work to help us help you. If you have any issues outstanding in JIRA please take a few moments to review them and gather the requested data or re-test with git MASTER. If no additional details can be determined please re-test you issue with Master and report those on your JIRA too. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org:81/key.txt (AB93356707C76CED)

3 3

One way audio only in a specified context
by Mr.Gab 06 May '14

06 May '14

Hello everybody, I have a problem with a FS installed on the cloud (configured following Freeswitch EC2 Wiki). Everything is ok except a one way audio behaviour that i have only if i transfer an outbound call. If I transfer an inbound call I don't have this problem. I'll explain the scenario with a little schema INBOUND CALL -> Extension 100 -> transfer to 101 WORKS. Extension 100 -> OUTBOUND CALL -> transfer to 101 ONE WAY AUDIO. I have one snom 300 (extension 100) and 3 Gigaset A510IP behind a tomato router. Can someone suggest me some way to manage this issue? Thanks in advice crazygabry

2 3

Voice channels in recorded file is out of sync
by Jayesh Nambiar 30 Apr '14

30 Apr '14

Hi, This problem has been reported several times in JIRA as well as lists without a proper solution. The problem is that the A-Leg runs a little ahead than the B-Leg in the recording only. The actual conversation is clear. I have following inputs to it: 1) This happens when the calls are made from a specific network and for calls that go beyond 20 minutes. 2) I tried this on Ubuntu 12.04 as well as Ubuntu 13.10, the problem still remains the same. 3) I've tried this with multiple version of freeswitch and the latest one I tested with is the following version: Version 1.5.12b+git~20140419T062924Z~f408dae2af~64bit (git f408dae 2014-04-19 06:29:24Z 64bit). In all versions the problem is consistent. 4) I tried with suppress-cng=true as well as false but the problem is still there. 5) I've added parameters like "bridge_generate_comfort_noise=true" before bridge. 6) I've tried with "record_waste_resources=true" before calling record_session. 7) I've tried with "record_fill_cng=true" before calling record_session. None of these helped me in solving this problem. I understand that it is something in the network probably, but I'll be greatful if anyone can point me for what to check on the network!! When can freeswitch record_session introduce such problem so that I can start looking into those directions. The endpoint being used was Grandstream GXP-1400. Is there a specific condition under which the freeswitch recording channels can go out of sync?? Because it starts well but suddenly when the conversation reaches 15-20 minutes the overlapping starts; meaning the channels start going out of sync !! Any help or pointers will be of great help. Thanks, --- Jayesh

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