August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Voice prompt cannot be played in FreeSWITCH(No sound)
by david.fu＠oocl.com 11 Aug '15

11 Aug '15

Hi All, We are new to FreeSWTICH and are trying to play voice prompt using dial plan in FreeSWITCH. However, we couldn't hear any sound. Would you please give us some hints ? We have shown the SIP log as belows. Thanks. Best Regards, David SIP Logs: ------------------------------------------------------------------------ 2015-08-11 12:54:46.449941 [DEBUG] sofia.c:6715 Channel sofia/internal/10810609(a)146.222.181.146 entering state [ready][200] 2015-08-11 12:54:46.749969 [DEBUG] switch_rtp.c:6462 Correct audio ip/port confirmed. recv 409 bytes from tcp/[146.222.181.146]:36548 at 12:54:48.462120: ------------------------------------------------------------------------ OPTIONS sip:146.222.181.242:5060 SIP/2.0 Via: SIP/2.0/TCP 146.222.181.146:5060;branch=z9hG4bK13f66bc21f08 From: <sip:146.222.181.146>;tag=1154629128 To: <sip:146.222.181.242> Date: Tue, 11 Aug 2015 05:09:01 GMT Call-ID: 12c4da00-5c91836d-13eb-92b5de92(a)146.222.181.146 User-Agent: Cisco-CUCM8.5 CSeq: 101 OPTIONS Contact: <sip:146.222.181.146:5060;transport=tcp> Max-Forwards: 0 Content-Length: 0 ------------------------------------------------------------------------ send 731 bytes to tcp/[146.222.181.146]:36548 at 12:54:48.462394: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TCP 146.222.181.146:5060;branch=z9hG4bK13f66bc21f08;rport=36548 From: <sip:146.222.181.146>;tag=1154629128 To: <sip:146.222.181.242>;tag=02ZDZ4QQvXKtg Call-ID: 12c4da00-5c91836d-13eb-92b5de92(a)146.222.181.146 CSeq: 101 OPTIONS Contact: <sip:146.222.181.242> User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20150810T224801Z~28ae331c93~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2015-08-11 12:54:49.029981 [DEBUG] switch_ivr_play_say.c:1805 done playing file /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav EXECUTE sofia/internal/10810609(a)146.222.181.146 sleep(500) 2015-08-11 12:54:49.549962 [NOTICE] switch_core_state_machine.c:315 sofia/internal/10810609(a)146.222.181.146 has executed the last dialplan instruction, hanging up. 2015-08-11 12:54:49.549962 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/internal/10810609(a)146.222.181.146 [CS_EXECUTE] [NORMAL_CLEARING] 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/10810609(a)146.222.181.146) State EXECUTE going to sleep 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/10810609(a)146.222.181.146) Running State Change CS_HANGUP 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/10810609(a)146.222.181.146) Callstate Change ACTIVE -> HANGUP 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/10810609(a)146.222.181.146) State HANGUP 2015-08-11 12:54:49.549962 [DEBUG] mod_sofia.c:431 Channel sofia/internal/10810609(a)146.222.181.146 hanging up, cause: NORMAL_CLEARING 2015-08-11 12:54:49.549962 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/10810609(a)146.222.181.146 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:60 sofia/internal/10810609(a)146.222.181.146 Standard HANGUP, cause: NORMAL_CLEARING 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/10810609(a)146.222.181.146) State HANGUP going to sleep 2015-08-11 12:54:49.549962 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/10810609(a)146.222.181.146) State Change CS_HANGUP -> CS_REPORTING 2015-08-11 12:54:49.569947 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/10810609(a)146.222.181.146) Running State Change CS_REPORTING 2015-08-11 12:54:49.569947 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/10810609(a)146.222.181.146) State REPORTING IMPORTANT NOTICE Email from OOCL is confidential and may be legally privileged. If it is not intended for you, please delete it immediately unread. The internet cannot guarantee that this communication is free of viruses, interception or interference and anyone who communicates with us by email is taken to accept the risks in doing so. Without limitation, OOCL and its affiliates accept no liability whatsoever and howsoever arising in connection with the use of this email. Under no circumstances shall this email constitute a binding agreement to carry or for provision of carriage services by OOCL, which is subject to the availability of carrier's equipment and vessels and the terms and conditions of OOCL's standard bill of lading which is also available at http://www.oocl.com.

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Display live calls in web pages
by John Nash 10 Aug '15

10 Aug '15

Is there any way of storing "live" calls (Ongoing calls) in DB to a specific table? where we can fetch these records and display in web page? I planned to store calls using some lua script hooks but cli command show calls already shows calls which we can use?

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Error msg "Invalid codec VP8!
by Alexander Haugg 10 Aug '15

10 Aug '15

Hi All, I'm using the actual freeswitch on Ubuntu in version 1.7.0+git~20150807... I added the VP8 codec in the vars.xml for global_codec_prefs and for outbound_codec_prefs. WS on Port 5066 is enabled and I use the "sipml5 demo client" as client. Normal audio calls works successfully (opus) but I get the error message in the CLI " [ERR] switch_core_codec.c:665 Invalid codec VP8! [ERR] switch_core_media.c:2671 Can't load codec? Mod_vp8 is build and in the modules.conf.xml enabled What can I do that the freeswitch allow the VP8 Codec with SIP? Thanks a lot!

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Curl xml handler
by Luis Daniel Lucio Quiroz 10 Aug '15

10 Aug '15

Just wondering if some one has a base example to start on.

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ESL library for the Dart programming language
by Kim Rostgaard Christensen 10 Aug '15

10 Aug '15

Hi list I've just uploaded my event socket library for the Dart language to their package repository. It is located at https://pub.dartlang.org/packages/esl And the sources (and the issue tracker) is located at https://github.com/rostgaard/libesl-dart Any feedback/issue/feature requests any of you may provide is greatly appreciated. Best /Kim

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Presence with shared database
by Adam Kuśmirek 08 Aug '15

08 Aug '15

Hello Everyone, We are working on platform with several Freeswitch nodes and shared Freeswitch database accessed with ODBC. It works great but we have a problem with presence. Scenario Phone A is registered on node 1 Phone A subscribes to phone B presence. It sends subscribe to Freeswitch node 1. Phone C calls phone B. It sends Invite to Freeswitch node 2. Freeswitch that is running on node 2 does not send Notify to phone A. If phone C sends Invite to Freeswitch node 1 everything is OK. Can anyone help ?

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Re: [Freeswitch-users] How to track call changes while running javascript?
by Thomas 07 Aug '15

07 Aug '15

They both don't change/fire. :/ It's as if the JS blocks everything about that call until it terminates. The hangup does not even register in the Freeswitch log until after the script finishes. Tried it outside execute_on_originate with the normal javascript application in the XML dialplan - same effect. The whole process seems to be a single thread - which would make sense for most use cases, unfortunately just not this one. :P Is there a way to fork the JS execution into the background? Something similar to setTimout() maybe? Or even better execute a dialplan app from JS that allows outside events to catch up? I tried the sleep application instead of JS msleep, but that didn't help either. On 01.08.2015 22:05, Ken Rice wrote: > You are looking for session ready or a hanguk up hook callback see confluence for details > > Sent from my iPhone > >> On Aug 1, 2015, at 3:53 PM, Thomas <lists(a)virtues.net> wrote: >> >> Does anyone know if there is a way to track changes to the call state, most notably a hangup, while executing >> Javascript? We're using JS to set up a call on a device connected to Freeswitch through portaudio. That call is >> initiated with sip/sofia executing the JS in the dialplan. If the SIP leg hangs up while the JS is running, nothing >> happens until the JS terminates. >> >> I wanted to check from the script, but haven't found a way to do that. "session.state" doesn't change, hangup hook >> doesn't fire, and without setTimeout or something similar I didn't find a way to "background" the JS. >> >> Is there any way to detect a hangup during JS execution? >> >> Cheers! >> >> Thomas >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org

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FreeSWITCH Friday FreeForAll Reminder!
by krice＠freeswitch.org 07 Aug '15

07 Aug '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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cannot setup srtp - Not Acceptable here
by support＠vip-consult.co.uk 07 Aug '15

07 Aug '15

when using jitsi and srtp i get SIP/2.0 488 Not Acceptable here tried few other sip clients and all return the same error the only thing I noticed is that although fs is set tosrtp at line 510 freeswitch send sip invite with m=audio 17672 RTP/AVP 96 9 0 8 103 3 101 13 instead of m=audio 17672 RTP/SAVP 96 9 0 8 103 3 101 13 https://pastebin.freeswitch.org/24346 https://www.dropbox.com/sh/hho493y2g9qykpq/AACvUjG9iT3oEvJvSg2KcatAa?dl=0 Kind Regards Chris

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srtp setup
by support＠vip-consult.co.uk 07 Aug '15

07 Aug '15

anything weird in the logs ?

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