- FreeSWITCH-users - FreeSWITCH Mailing Lists

488 Not Acceptable Media during blind call transfer in bypass media mode
by ilyakn 10 Jan '15

10 Jan '15

Hi All, I've run into a problem during call transfer. I have a CISCO switch calling Freeswitch that set bypass_media to true and makes a bridge to IVR platform (its actually a VXML browser). When IVR platform then tries to transfer a call using REFER the Freeswitch sends INVITE to the gateway in order to perform call transfer but Media part inside SDP is empty - as a result gateway returns 488 Not Acceptable Media and transfer fails. Is it possoble to make Freeswitch to send meaningful SDP inside INVITE? I would expect it to copy SDP from original incoming call - but this does not happen. Thanks, Ilya -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/488-Not-Acceptable-Media-duri… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Launch a script on errors in freeswitch log file
by Aqs Younas 09 Jan '15

09 Jan '15

Hi, How can i launch a script when there are error logs in freeswitch log file. Actually i am using mod_shout as MOH with mod_local_stream, when someone enters a url of stream that is not responding i see logs all these. 2015-01-09 19:38:39.338003 [ERR] mod_local_stream.c:214 Can't open shout:// 174.37.194.139:8316 2015-01-09 19:38:41.337988 [ERR] switch_core_file.c:149 Invalid file format [shout] for [174.37.194.139:8316]! 2015-01-09 19:38:42.337988 [ERR] mod_local_stream.c:214 Can't open shout:// s9.voscast.com:7584 2015-01-09 19:38:42.337988 [ERR] switch_core_file.c:149 Invalid file format [shout] for [s9.voscast.com:7584]! I want to run an external script to remove that file or urls when there are errors logs in freeswitch. But i don't know how to trigger that script on errors in freeswitch log. Any help would be much appreciated. Thanks.

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 09 Jan '15

09 Jan '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

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Dialplan sip chat with SIPml5: client receives message, but freeswitch reports 'Error! Message Not Sent'
by Question Answer 09 Jan '15

09 Jan '15

Hi, I have successfully configured a SIPml5 client in Chrome to communicate with a local FreeSWITCH server. It is possible to successfully make calls, and use most applications in the freeswitch dial plan. Problem: I cannot figure out how to successfully send a custom SIP message from FreeSWITCH [via either the CLI or the dialplan] to the web client. The client actually *receives* the message, but FreeSWITCH then hangs for about 10s, times out and reports: 'Error! Message Not Sent Sent'. I run the following command from the CLI: freeswitch> chat sip|1001@192.168.59.103|1000@172.17.0.19|test test test send 764 bytes to ws/[192.168.59.3]:51933 at 09:48:10.190253: ------------------------------------------------------------------------ MESSAGE sip:1000@192.168.59.3:51933 SIP/2.0 Via: SIP/2.0/WS 172.17.0.19:5066;branch=z9hG4bKFpeyg98ZZ240r Route: <sip:1000@192.168.59.3:51933>;rtcweb-breaker=yes;transport=ws Max-Forwards: 70 From: "1001" <sip:1001@192.168.59.103>;tag=U9prcNXypBBUr To: <sip:1000@192.168.59.3:51933>;rtcweb-breaker=yes;transport=ws Call-ID: 5b3c7d7c-8149-422c-9f0f-f1b3dd3661f2 CSeq: 70061165 MESSAGE Contact: <sip:172.17.0.19> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150106T091916Z~f48ec61d54~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: text/html Content-Length: 14 X-FS-Sending-Message: 4a972b6a-3c72-4ba1-b7a4-de2d07760380 test test test ------------------------------------------------------------------------ [~10 seconds pass...] Error! Message Not Sent ... freeswitch> -------------------------------------------------------------------------------- But on the client side, in the Chrome developer console, I can see the following log messages: SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage 2015-01-09 11:37:09.647SIPml-api.js?svn=222:1 recv=MESSAGE sip:1000@192.168.59.3:52190 SIP/2.0 Via: SIP/2.0/WS 172.17.0.19:5066;branch=z9hG4bKKtK1QNcem6XBQ From: "1001"<sip:1001@192.168.59.103>;tag=29gZHme856Fmc To: <sip:1000@192.168.59.3:52190>;rtcweb-breaker=yes;transport=ws Contact: <sip:172.17.0.19> Call-ID: 210f6e1f-0415-450b-bbb3-7725a6a37434 CSeq: 70061168 MESSAGE Content-Type: text/html Content-Length: 14 Route: <sip:1000@192.168.59.3:52190> Max-Forwards: 70 User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150106T091916Z~f48ec61d54~64bit Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE Supported: timer,path,replaces X-FS-Sending-Message: 4a972b6a-3c72-4ba1-b7a4-de2d07760380 test test test 2015-01-09 11:37:09.647SIPml-api.js?svn=222:1 State machine: tsip_dialog_generic_Started_2_Incoming_X_iMessage 2015-01-09 11:37:09.649SIPml-api.js?svn=222:1 ==stack event = i_new_message XXX test test test 2015-01-09 11:37:09.649call.htm:964 test test test ---------------------------------------------------------------------------------- That is, the message is correctly received by the client, but FreeSWITCH doesn't seem to 'realise' this. Perhaps I need to be sending some kind explicit confirmation back from the simpml5 client application? Or perhaps there is some kind of mismatch with the SIP profile? I have set the freeswitch log level to Debug, but the above is the only message I see, which is not particularly helpful. Any insights would be appreciated!

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Event Socket: CHANNEL_DESTROY for a instantly Failing Call not containing custom channel variables
by Joshua Gigg 09 Jan '15

09 Jan '15

Hi, I'm using the Inbound Event Socket to listen to events. When I try to originate a call that fails instantly, the CHANNEL_DESTROY event I receive (which is the first event for that Call) does not seem to contain the custom variables I passed in the origination command. E.g. - Subscribe to all events - originate {myFirstVar=bla}sofia/external/ &park; This errors straight away in the CLI, but event wise: {"Event-Name":"CHANNEL_DESTROY","Core-UUID":"0bdb8ae0-95be-11e4-a629-7deec23cf31f","FreeSWITCH-Hostname":" test-fs1.example.com","FreeSWITCH-Switchname":"test-fs1.example.com","FreeSWITCH-IPv4":"203.0.113.1","FreeSWITCH-IPv6":"::1","Event-Date-Local":"2015-01-09 10:07:47","Event-Date-GMT":"Fri, 09 Jan 2015 10:07:47 GMT","Event-Date-Timestamp":"1420798067526747","Event-Calling-File":"switch_core_session.c","Event-Calling-Function":"switch_core_session_perform_destroy","Event-Calling-Line-Number":"1465","Event-Sequence":"43492","Channel-State":"CS_NONE","Channel-Call-State":"DOWN","Channel-State-Number":"0","Channel-Name":"N/A","Unique-ID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Call-Direction":"outbound","Presence-Call-Direction":"outbound","Channel-HIT-Dialplan":"false","Channel-Call-UUID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Answer-State":"ringing","variable_direction":"outbound","variable_is_outbound":"true","variable_uuid":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","variable_call_uuid":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","variable_session_id":"420"}Content-Length: 880 Content-Type: text/event-json {"Event-Name":"CHANNEL_STATE","Core-UUID":"0bdb8ae0-95be-11e4-a629-7deec23cf31f","FreeSWITCH-Hostname":" test-fs1.example.com","FreeSWITCH-Switchname":"test-fs1.example.com","FreeSWITCH-IPv4":"203.0.113.1","FreeSWITCH-IPv6":"::1","Event-Date-Local":"2015-01-09 10:07:47","Event-Date-GMT":"Fri, 09 Jan 2015 10:07:47 GMT","Event-Date-Timestamp":"1420798067526747","Event-Calling-File":"switch_channel.c","Event-Calling-Function":"switch_channel_perform_set_running_state","Event-Calling-Line-Number":"2193","Event-Sequence":"43493","Channel-State":"CS_DESTROY","Channel-Call-State":"DOWN","Channel-State-Number":"0","Channel-Name":"N/A","Unique-ID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Call-Direction":"outbound","Presence-Call-Direction":"outbound","Channel-HIT-Dialplan":"false","Channel-Call-UUID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Answer-State":"ringing"}Content-Length: 608 Content-Type: text/event-json Ideally, I shouldn't be making calls to numbers that would fail like this, but shouldn't I still be getting the variables I send across? -- Joshua Gigg

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P-Asserted-Identity: <tel:+1404XXXXXXX>
by George F. Phelps 08 Jan '15

08 Jan '15

Inside my dialplan, I have: <action application="set" data="sip_h_P-Asserted-Identity=tel:+1404XXXXXXX"/> In the INVITE SIP trace, I see: P-Asserted-Identity: <tel:+1404XXXXXXX> I.e., the data portion of my original character string - "tel:+1404XXXXXXX" - has been encapsulated by less-than ("<") and greater-than (">") characters. How do I code my dialplan to get. P-Asserted-Identity: tel:+1404XXXXXXX .as opposed to the current: P-Asserted-Identity: <tel:+1404XXXXXXX> Thanks, George

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how can i do stream piping in freeswitch
by Aqs Younas 08 Jan '15

08 Jan '15

Currently i am playing a stream with mod_shout and this is my default xml. <context name="default"> <extension name="global-intercept"> <condition field="destination_number" expression="^14049002000$"> <action application="answer"/> <action application="playback" data="shout:// online.radiodifusion.net:8024/"/> </condition> </extension> </context> Every time a user makes a call for stream it opens a separate connection with the stream provider. If 100 users dials this number, there would be 100 connections with stream provider listening to same stream, means more rtp packets containing same data for different users. What i want is, if a user is listening to stream then other users must share the same listening connection that the first user is opened, instead of creating a separate connection with stream provider for same stream. Someone told me this is possible in asterisk, so there must be a way in freeswitch. How can i do this.? Any help would be much appreciated. Thanks

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Wrong ice candidate (internal ip) when using mod_verto
by Adam Ben-Ayoun 08 Jan '15

08 Jan '15

Hi, I am experimenting with mod_verto and using the demo to call to a conference on my server (logs on FreeSwitch seems ok generally), the issue is that I am not receiving/sending any audio (I would expect to hear moh at this point), I suspect this is because FreeSwitch sends an ice host candidate set to the internal ip instead of the external, here is the SDP: o=FreeSWITCH 1420641299 1420641300 IN IP4 10.245.43.195 s=FreeSWITCH c=IN IP4 10.245.43.195 t=0 0 a=msid-semantic: WMS Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br m=audio 26060 RTP/SAVPF 111 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 maxplaybackrate=48000; minptime=10 a=rtpmap:126 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=fingerprint:sha-256 0D:BB:87:E0:B6:8A:AE:0D:32:3F:54:9F:36:FF:9B:89:28:AB:B4:17:BC:95:0D:2E:48:0C:33:30:78:52:5F:24 a=rtcp-mux a=rtcp:26060 IN IP4 10.245.43.195 a=ssrc:1435389439 cname:TfPkXSQ755IUEU1s a=ssrc:1435389439 msid:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a0 a=ssrc:1435389439 mslabel:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a=ssrc:1435389439 label:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Bra0 a=ice-ufrag:hQuEgraFWcUo2lGL a=ice-pwd:qmWe5Cb0xJL2tDkAjF4pW5UJ a=candidate:0460306999 1 udp 659136 10.245.43.195 26060 typ host generation 0 Using Wireshark I see that the client is trying to send packets to 10.245.43.195 which obviously fails.. How can I fix this? Thanks, Adam

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Register->From->Caller ID
by George F. Phelps 08 Jan '15

08 Jan '15

My current gateway REGISTER packet has a certain FROM header, as shown below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: <sip:+1404XXXXXXX@66.33.147.150>;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150> Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: <sip:gw+switch2voip.us@54.174.255.168:5080;transport=udp;gw=switch2voip.us> . . . What do I set/change in my gateway configuration to add a character string "1404XXXXXXX" to the FROM header? As shown in red below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: "1404XXXXXXX" <sip:+1404XXXXXXX@66.33.147.150>;tag=FQag8jSNpac9c To: <sip:+1404XXXXXXX@66.33.147.150> Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: <sip:gw+switch2voip.us@54.174.255.168:5080;transport=udp;gw=switch2voip.us> . . . I am already setting the following values (in my gateway definition): <param name="from-user" value="1404XXXXXXX"/> <param name="caller-id-in-from" value="true"/> Thanks, George

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Fwd: OverTime RAM usage keep inceasing
by indra sena 08 Jan '15

08 Jan '15

Hi Team, I have observed that overtime RAM utilization/usage is keep increasing maybe due to memory leaks ? when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 hours load test and oberserved now it is using almost 11GB/12GB starts swap also increasing now. Do you have config changes to fix or to take as work around ? How can we reduce RAM utilization overtime ? Could you please provide some solution for this ? Thanks & Regards, Indra.

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