- FreeSWITCH-users - FreeSWITCH Mailing Lists

Testing/Automation Meeting Tuesday July 14th 2PM Central
by Brian West 10 Jul '15

10 Jul '15

Where: https://cantina.freeswitch.org/conf/ Topic: Discussion about Automated QA and Unit testing framework. Reply to this thread and we can work on our agenda prior to the call. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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freeswitch and fusionpbx
by Joli Martinez 10 Jul '15

10 Jul '15

I am running CentOS 6.5 with freeswitch installed on /usr/local/freeswitch. This was working for a while, I decided to install fusionPBX to get a gui interface. I ran through the install and I am able to login to FusionPBX but it seems it is not tied to the Freeswitch install. It does not see my current extensions and when I add extensions on Fusion they do not show up in Freeswitch. Freeswitch is installed with MySql. Where do I go to add my current freeswitch install to fusionpbx? Thanks

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Bypass media prevents bridge from blocking LUA script execution
by Ričardas Stoma 10 Jul '15

10 Jul '15

I use LUA script to bridge calls but now i have problem with bypass media. Let's say i have this LUA script: session:execute("set", "bypass_media=true"); session:execute("bridge", "sofia/external/123@192.168.0.100:5061"); freeswitch.consoleLog("info", "This is my message\n"); When bypass_media is enabled, log message "This is my message" appears once legB answers call. If bypass_media is disabled then log message appears when call ends. In other words bypass_media prevents "bridge" from blocking script execution. Is there a way to block script execution until call ends when bypass_media is enabled?

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Preferred Crypto for outbound calls
by Prashanth Devarajappa 09 Jul '15

09 Jul '15

Hello, How do I set my preferred crypto suite to be used for SRTP outbound call ? I am running FS 1.4.18. In old version of FS(1.2.3), I could do that using sip_secure_media=AES_CM_128_HMAC_SHA1_80. But this variable doesn't exist anymore, by the looks of it. Regards Prashanth

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No Video in Media Proxy mode
by Bryan 09 Jul '15

09 Jul '15

Hi, I'm testing FreeSWITCH with Android Linphone. I found the Video stream was not forwarded by FS. Audio is OK. Any Ideas? Or this is a bug? The test envrionmet as following: Android Linphone < -- > Wifi Router < -- > FS < -- >PC Linphone. The FS Version is the Latest Windows MSI package. 2015-07-03 14:31:57.718750 [NOTICE] switch_channel.c:1055 New Channel sofia/inte rnal/1009(a)172.16.1.115 [c8a19d44-3084-4539-b817-3276fd7ec43d] 2015-07-03 14:31:57.750000 [INFO] mod_dialplan_xml.c:635 Processing 1009 <1009>- >1007 in context default 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1 execut e_extension::dx XML features 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2 record _session::C:/Program Files/FreeSWITCH/recordings/1009.2015-07-03-14-31-57.wav 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3 execut e_extension::cf XML features 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4 execut e_extension::att_xfer XML features 2015-07-03 14:31:57.781250 [NOTICE] switch_channel.c:1055 New Channel sofia/inte rnal/sip:1007@172.16.1.115:2887 [4cfdf0e6-7010-467b-97b3-20e4dc295422] 2015-07-03 14:31:57.843750 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:1 007@172.16.1.115:2887! 2015-07-03 14:31:57.843750 [NOTICE] mod_sofia.c:2107 Ring-Ready sofia/internal/1 009(a)172.16.1.115! 2015-07-03 14:31:57.859375 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/ internal/1009(a)172.16.1.115! 2015-07-03 14:32:00.312500 [NOTICE] sofia.c:7437 Channel [sofia/internal/sip:100 7@172.16.1.115:2887] has been answered 2015-07-03 14:32:00.312500 [NOTICE] switch_core_media.c:4405 sofia/internal/sip: 1007@172.16.1.115:2887 Starting Video thread 2015-07-03 14:32:00.343750 [NOTICE] switch_core_media.c:4405 sofia/internal/1009 @172.16.1.115 Starting Video thread 2015-07-03 14:32:00.343750 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/i nternal/1009(a)172.16.1.115] has been answered 2015-07-03 14:32:00.843750 [INFO] switch_rtp.c:5832 Auto Changing port from 61.1 83.139.155:64131 to 172.16.15.111:7078 2015-07-03 14:32:23.750000 [NOTICE] sofia.c:952 Hangup sofia/internal/sip:1007@1 72.16.1.115:2887 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-07-03 14:32:23.750000 [NOTICE] switch_ivr_bridge.c:1608 Hangup sofia/intern al/1009(a)172.16.1.115 [CS_EXECUTE] [NORMAL_CLEARING] 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/ internal/sip:1007@172.16.1.115:2887) Ended 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close Channel sof ia/internal/sip:1007@172.16.1.115:2887 [CS_DESTROY] 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 3 (sofia/ internal/1009(a)172.16.1.115) Ended 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close Channel sof ia/internal/1009(a)172.16.1.115 [CS_DESTROY] Tony

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mod_rtmp onStatus messages
by Paul Cuttler 09 Jul '15

09 Jul '15

I was wondering about the onStatus messages that are sent by mod_rtmp to Flash Player. The RTMP Spec (see page 38 of http://wwwimages.adobe.com/content/dam/Adobe/en/devnet/rtmp/pdf/rtmp_specif…) says that the transaction ID for onStatus messages is supposed to be 0, but mod_rtmp uses 1. It seems to work fine, but I wanted to check if anyone knew if it might lead to problems not by following the spec. This is in rtmp_sig.c, for example in rtmp_i_play. regards, Paul

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Fwd: sofia_reg_handle_register_token - exp possibly always false
by Podrigal, Aron 09 Jul '15

09 Jul '15

---------- Forwarded message ---------- From: Podrigal, Aron <aronp(a)guaranteedplus.com> Date: Tue, Jul 7, 2015 at 7:24 PM Subject: sofia_reg_handle_register_token - exp possibly always false To: freeswitch-dev <freeswitch-dev(a)lists.freeswitch.org> Hi, According to grep the function sofia_reg_handle_register_token is called from a single place src/mod/endpoints/mod_sofia/sofia_reg.c:2305 <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/en…> where the argument *v_event* would always be NULL at that time. So I'm wondering what the line at src/mod/endpoints/mod_sofia/sofia_reg.c:1297 <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/en…> if (v_event && *v_event) pres_on_reg = switch_event_get_header(*v_event, "send-presence-on-register"); is suppose to do, won't this always evaluate to false? -- Aron Podrigal - //Be happy :-) -- Aron Podrigal - //Be happy :-)

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Fax problem
by Alexey Melnichuk 09 Jul '15

09 Jul '15

FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~32bit (git b95362f 2015-07-03 16:42:15Z 32bit) I think I found bug in spandsp. `t4_image_put_handler_t` defined as (__cdecl) typedef int (*t4_image_put_handler_t)(void *user_data, const uint8_t buf[], size_t len); But `tXX_decode_put` functions defined as `__stdcall`. So when I do s->image_put_handler = (t4_image_put_handler_t) t4_t6_decode_put; ... s->image_put_handler(...) I got AV. After I declare t4_image_put_handler_t as __stdcall problem solved. (Alse need update declaretion to `pre_encoded_put` function) But I start getting heap corruption on `span_free((char *) s->tiff.file);` Problem is that it uses `strdup` function with different allocator. So I just copy function to `alloc.c` SPAN_DECLARE(char *) span_strdup(const char *str) { char *sdup; size_t len = strlen(str) + 1; sdup = (char *) span_alloc(len); memcpy(sdup, str, len); return sdup; } And use it as ds->tiff.file = span_strdup(file);

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extra Bye message from Freeswitch
by Jurijs Ivolga 09 Jul '15

09 Jul '15

Hi! I started to run some performance tests against Freeswitch and in 5-10% tests are failing because of Freeswitch sending extra bye message. I would like to add that I got such messages on very low load, 1 new call in 5 seconds and first fail was after 9th call. Here how call looks like in 90-95% of time: SIPp UAC Freeswitch |(1) INVITE | |------------------>| |(2) 100 (optional) | |<------------------| |(3) 200 | |<------------------| |(4) ACK | |------------------>| |(5) BYE | |------------------>| |(6) 200 | |<------------------| Below you can see how call looks like in 5-10% of time. Somehow Freeswitch replies on bye message from Sipp with extra bye(around 5-10% of all calls) and it creates a mess: SIPp UAC Freeswitch |(1) INVITE | |------------------>| |(2) 100 (optional) | |<------------------| |(3) 200 | |<------------------| |(4) ACK | |------------------>| |(5) BYE | |------------------>| |(6) BYE | |<------------------| |(7) 200 | |------------------>| |(8) 200 | |<------------------| I'm not sure, but I think such behavior is not expected. Maybe somebody can let me know if it is proper Freeswitch behavior and if it is, why such behavior only happens with 5-10% of calls. Below you can find full sip trace, for a call with extra bye message. U 2015/07/09 11:07:07.269998 sippserver:5060 -> freeswitchserver:5060 INVITE sip:service@freeswitchserver:5060 SIP/2.0. Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0. From: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. To: service <sip:service@freeswitchserver:5060>. Call-ID: 9-11560@sippserver. CSeq: 1 INVITE. Contact: sip:sipp@sippserver:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 sippserver. s=-. c=IN IP4 sippserver. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 2015/07/09 11:07:07.271005 freeswitchserver:5060 -> sippserver:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0. From: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. To: service <sip:service@freeswitchserver:5060>. Call-ID: 9-11560@sippserver. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit. Content-Length: 0. . # U 2015/07/09 11:07:07.276784 freeswitchserver:5060 -> sippserver:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0. From: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. To: service <sip:service@freeswitchserver:5060>;tag=S3Kvy09a3BrtS. Call-ID: 9-11560@sippserver. CSeq: 1 INVITE. Contact: <sip:service@freeswitchserver:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 168. Remote-Party-ID: "service" <sip:service@freeswitchserver >;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1436408221 1436408222 IN IP4 freeswitchserver. s=FreeSWITCH. c=IN IP4 freeswitchserver. t=0 0. m=audio 21006 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2015/07/09 11:07:07.277349 sippserver:5060 -> freeswitchserver:5060 ACK sip:service@freeswitchserver:5060 SIP/2.0. Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-5. From: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. To: service <sip:service@freeswitchserver:5060>;tag=S3Kvy09a3BrtS. Call-ID: 9-11560@sippserver. CSeq: 1 ACK. Contact: sip:sipp@sippserver:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 2015/07/09 11:07:07.278861 sippserver:5060 -> freeswitchserver:5060 BYE sip:service@freeswitchserver:5060 SIP/2.0. Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-7. From: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. To: service <sip:service@freeswitchserver:5060>;tag=S3Kvy09a3BrtS. Call-ID: 9-11560@sippserver. CSeq: 2 BYE. Contact: sip:sipp@sippserver:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 2015/07/09 11:07:07.278931 freeswitchserver:5060 -> sippserver:5060 BYE sip:sipp@sippserver:5060 SIP/2.0. Via: SIP/2.0/UDP freeswitchserver;rport;branch=z9hG4bKF5jjXD7vrF3Qp. Max-Forwards: 70. From: service <sip:service@freeswitchserver:5060>;tag=S3Kvy09a3BrtS. To: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. Call-ID: 9-11560@sippserver. CSeq: 77878549 BYE. User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . # U 2015/07/09 11:07:07.279599 sippserver:5060 -> freeswitchserver:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP freeswitchserver;rport;branch=z9hG4bKF5jjXD7vrF3Qp. From: service <sip:service@freeswitchserver:5060>;tag=S3Kvy09a3BrtS. To: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. Call-ID: 9-11560@sippserver. CSeq: 77878549 BYE. Contact: <sip:sippserver:5060;transport=UDP>. Content-Length: 0. . # U 2015/07/09 11:07:07.280338 freeswitchserver:5060 -> sippserver:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-7. From: sipp <sip:sipp@sippserver:5060>;tag=11560SIPpTag009. To: service <sip:service@freeswitchserver:5060>;tag=S3Kvy09a3BrtS. Call-ID: 9-11560@sippserver. CSeq: 2 BYE. Content-Length: 0. . I will appreciate any help! Thank you!

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No Bridge Media
by Vishal Sharma 09 Jul '15

09 Jul '15

Hi, When I use following commnd on fs_cli, I can hear ring sound or any message telecome provider play on destination number originate freetdm/outgoing/r/09899790092 &bridge(freetdm/outgoing/r/09899488723) but when I use one of following commands freeswitch@internal> originate {bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge (freetdm/outgoing/r/09899488723) freeswitch@internal> originate {bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge({bridge_early_media=true}freetdm/outgoing/r/09899488723) or when I bridge a call which is received on FS, I don't get bridge media. Is there any perticular config required, or it's bug.. Regards, Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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