January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Doc-Sprint Friday 16 January 2015
by Areski 12 Jan '15

12 Jan '15

Hi Everyone, We are organizing an other Sprint to work on FreeSWITCH Documentation on *Friday 16 January 2015 at 10am CT*. It will be 4 hours long but you can join for less time. The Doc-sprint will focus on migrating the remaining pages from old FS Wiki (https://wiki.freeswitch.org) to Confluence ( https://freeswitch.org/confluence) We will use this IRC channel during the sprint: #freeswitch-docs and we will be tracking our work on a spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYa… to avoid working on the same content. So during the sprint, please change the page "Status" you are working on to "Editing" with your name next to it. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documenta… Let us know if you have any question. We hope to get a maximum number of people signed up! So, who is in? -- Kind regards, /Areski <http://www.linkedin.com/in/areski>

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Re: [Freeswitch-users] REg IVR and codec negotiation
by indra sena 12 Jan '15

12 Jan '15

Hi , I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, GISR..

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Set From User in SIP request
by jeremy ardley 12 Jan '15

12 Jan '15

Hi, I have a DID provider with multiple incoming numbers. I can send outgoing calls on particular DID numbers by sending a SIP request with the From header formatted as From: dddddddddd <sip:dddddddd@sip.provider.com>;tag=yyy Where the From name is the DID in full national number form (10 digit) and the From user is the 8 digit account number. Each unique DID has a matching unique account number. I have internal extensions mapped to different DID numbers. I need a mechanism to feed different DID number and account number pairs to the gateway for each outgoing call. I hope to do that via the internal extension xml files e.g. <include> <user id="200"> <params> <param name="password" value="xxxxxx"/> <param name="vm-password" value="200"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="200"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Michael J Fox"/> <variable name="effective_caller_id_number" value="200"/> <variable name="outbound_caller_id_name" value="0000000001"/> <variable name="outbound_caller_id_number" value="0000000001"/> <variable name="outbound_caller_from_user" value="01234567"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> The question is how to pass the new variable outbound_caller_from_user to the gateway and get it substituted in the user field of the SIP From header? I have a secondary issue in that I can set the outbound DID part of the >From Header but it has quotes around it From: "dddddddddd" <sip:dddddddd@sip.provider.com>;tag=yyy Which is undesirable and it would be helpful to suppress that in Freeswitch itself. Thanks,

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Call back through SIP trunk
by Sina Owolabi 10 Jan '15

10 Jan '15

Hi List! FreeSWITCHNewbie here. Please can I have some guidance on how to setup call back? I would like to be able to dial the DID attached to the SIP trunk Freeswitch is registered to, and then have freeSWITCH hang up the call and dial the caller id number back through any other SIP trunk FreeSWITCH Is registered with, but with the origination number set to the DID that the first call came through in the first place. Please is this possible just through the dial plan? Thanks for any help!

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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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