May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

DTMF Issue
by Richard Mace 05 Jun '14

05 Jun '14

Hi All, I have a FreeSWITCH server running behind a firewall and can successfully connect a Jitsi software client to the server remotely. I can dial 5000 from Jitsi and hear audio, but when I dial a DTMF digit,nothing happens. I have tried configuring Jitsi to use different DTMF methods, but none seem to work. It is a vanilla install of FreeSWITCH, Running on Debian 7 installed via the packages so the config files will be default. Any ideas? Thanks Richard

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usb modem setup
by Joli Martinez 02 Jun '14

02 Jun '14

I have a USB modem that I would like to connect it to Freeswitch as fax server. How can I get freeswitch to detect the modem? Thanks,

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FreeSWITCH 1.4.5 Released!
by Ken Rice 02 Jun '14

02 Jun '14

FreeSWITCH 1.4.5 has been released! Improvements include: Resolve issue with mod_sofia SIP subscriptions over UDP. Fixes and cleanup of mod_cepstral, mod_fifo, and other modules. The FreeSWITCH Team is currently doing a large review with Coverity and are correcting many potential issues. Patches for these things are also included. Tarball is available http://files.freeswitch.org/freeswitch-1.4.5.tar.gz See you @ Cluecon! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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503 Maximum Calls In Progress
by Miha 01 Jun '14

01 Jun '14

Hi, I am getting "503 Maximum Calls In Progress". I did some sofia.conf.xml configuration but still the same. Could some please point me to right direction what I am missing. tnx! UP 0 years, 5 days, 1 hour, 2 minutes, 55 seconds, 545 milliseconds, 798 microseconds FreeSWITCH (Version 1.2.16 git 3bbb60f 2013-12-15 03:45:56Z 64bit) is ready 266825 session(s) since startup 111 session(s) - peak 177, last 5min 117 0 session(s) per Sec out of max 60, peak 18, last 5min 6 1000 session(s) max min idle cpu 25.00/84.00 Current Stack Size/Max 240K/8192K  <param name="min-idle-cpu" value="25"/>  <param name="max-sessions" value="1000"/>  <param name="sessions-per-second" value="60"/>

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connecting flynumber to freeswitch
by Jamie McClelland 30 May '14

30 May '14

Hi all, I've purchased a flynumber (http://www.flynumber.com/) - which gives me a local mexico city number. It has an option to forward that to a SIP account. The flynumber configuration asks for the protocol (SIP), the host (talk.mayfirst.org) and the "Username/ext" (I've put 9999 - since the extension 9999(a)talk.mayfirst.org is our public conference line). It doesn't provide me with an option to enter a password to register with. When I call the number, I get the following in our console (see below). I think that's because I'm not explicitly allowing un-registered callers to dial that extension. Is that true? And if so - can someone point me to the right configuration/docs on how to enable that? I'm not sure this is something that should be set in our dial plan, or in sip_profile/internal/? Many thanks! jamie p.s. I would like to enable any un-authenticated SIP user to be able to call 9999(a)talk.mayfirst.org. b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/525560504865(a)46.19.209.14 [b1322da6-8e70-47d0-8910-ea2a319fd741] b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/525560504865(a)46.19.209.14 [BREAK] b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/525560504865(a)46.19.209.14 [BREAK] b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/525560504865(a)46.19.209.14) Running State Change CS_NEW b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [DEBUG] sofia.c:8340 sofia/internal/525560504865(a)46.19.209.14 receiving invite from 46.19.209.14:5060 version: 1.4.4 -1 64bit 2014-05-29 12:50:08.412870 [DEBUG] sofia.c:8507 IP 46.19.209.14 Rejected by acl "domains". Falling back to Digest auth. b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/525560504865(a)46.19.209.14 [BREAK] 2014-05-29 12:50:08.412870 [DEBUG] sofia.c:2033 detaching session b1322da6-8e70-47d0-8910-ea2a319fd741 b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:08.412870 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/525560504865(a)46.19.209.14) State NEW b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [WARNING] switch_core_state_machine.c:567 b1322da6-8e70-47d0-8910-ea2a319fd741 sofia/internal/525560504865(a)46.19.209.14 Abandone d b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [NOTICE] switch_core_state_machine.c:570 Hangup sofia/internal/525560504865(a)46.19.209.14 [CS_NEW] [WRONG_CALL_STATE] b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_channel.c:3216 Send signal sofia/internal/525560504865(a)46.19.209.14 [KILL] b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_session.c:1387 Send signal sofia/internal/525560504865(a)46.19.209.14 [BREAK] b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/525560504865(a)46.19.209.14) Running State Change CS_HANGUP b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_state_machine.c:730 (sofia/internal/525560504865(a)46.19.209.14) Callstate Change DOWN -> HANGUP b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_state_machine.c:732 (sofia/internal/525560504865(a)46.19.209.14) State HANGUP b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] mod_sofia.c:413 Channel sofia/internal/525560504865(a)46.19.209.14 hanging up, cause: WRONG_CALL_STATE b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_state_machine.c:58 sofia/internal/525560504865(a)46.19.209.14 Standard HANGUP, cause: WRONG_CALL_STATE b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_state_machine.c:732 (sofia/internal/525560504865(a)46.19.209.14) State HANGUP going to sleep b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/525560504865(a)46.19.209.14) State Change CS_HANGUP -> CS_REPORTING b1322da6-8e70-47d0-8910-ea2a319fd741 2014-05-29 12:50:18.492927 [DEBUG] switch_core_session.c:1387 Send signal sofia/internal/525560504865(a)46.19.209.14 [BREAK] -- May First/People Link Growing networks to build a just world http://www.mayfirst.org https://support.mayfirst.org OpenPGP Key: http://current.workingdirectory.net/pages/identity/ xmpp: jamie(a)im.mayfirst.org

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version 1.4.4 or 1.5.13b
by Shlomi Agiv 30 May '14

30 May '14

Hi, Whenever I try to get the latest stable release, I get version 1.5.13b. What is the git command to get the latest stable code? sorry in advance for the lack of git knowledge. Thanks, Shlomi

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ESL C# Accessviolation
by Ferenc Sárközy 30 May '14

30 May '14

Hi, My C# app uses ESL to interact with FS. It plays wav files via bgapi and receives and processes the events of the playbacks. The listening thread is started like ManagedESLTest program: ThreadPool.QueueUserWorkItem(new WaitCallback(getFsEventsInboundMode)); this thread ends up in a while loop where the events are received: while (eslConnection.Connected() == ESL_SUCCESS && !stopConnection) { eslEvent = eslConnection.RecvEventTimed(50); if (eslEvent != null) { onFreeswitcEvent(eslEvent ); } } onFreeSwitchEvent calls ends up in an other thread. bgapi commands is sent via the same eslConnection, I assume that the occasional Accessviolation at RecvEventTimed is caused by sending bgapi commands via the same eslConnection from an other thread. Could that be the cause? Other related questions: Is it safe to pass eslEvent to an other thread, or it should be serialized? If so is there a best practice to serialize? Is there some kind of event buffer in eslConnection, or the RecvEvent misses all the events which "arrived" before the call? Thank you in advance: Ferenc

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Event Socket Layer State Diagram
by Grant Bagdasarian 30 May '14

30 May '14

Hello, Is there a state diagram available which shows the state transitioning for all Channel Events and Channel States? Also, I've wrote a simple ESL application which listens for all events happening in FS during a single call, and I've been studying the Channel Events/States and the correlation between them. Very useful. I recommend every new ESL developer to do this. I do have a small question. When using the SendMSG method of ESLConnection to playback a file, the below sequence of events is generated. Now my question is: is there a way to send some unique string with the sendmsg command and have that one appear in all the other events which are shown below. I need to track the events which share a relationship. CHANNEL_EXECUTE and CHANNEL_EXECUTE_COMPLETE contain a unique Application-UUID, which is good and could suffice, but I haven't found a way where I can relate the sendmsg command back to these two events. I know there is the Unique-ID header, but I'd like every action to be uniquely identifiable. Is that possible or am I asking way too much? "Command": "sendmsg c48f66c0-e5be-11e3-a832-551ac922334e", "Event-Subclass": "c48f66c0-e5be-11e3-a832-551ac922334e", "call-command": "execute", "execute-app-name": "playback", "execute-app-arg": "/usr/src/freeswitch/sounds/basic/welcome.wav", "event-lock": "true", "Event-Name": "PRIVATE_COMMAND", "Event-Name": "CHANNEL_EXECUTE", "Application": "playback", "Application-Data": "/usr/src/freeswitch/sounds/basic/welcome.wav", "Application-UUID": "ca0df0a8-e5be-11e3-a83e-551ac922334e" "Event-Name": "PLAYBACK_START", "Playback-File-Path": "/usr/src/freeswitch/sounds/basic/welcome.wav", "Event-Name": "PLAYBACK_STOP", "Playback-File-Path": "/usr/src/freeswitch/sounds/basic/welcome.wav", "Playback-Status": "done" "Event-Name": "CHANNEL_EXECUTE_COMPLETE", "Application": "playback", "Application-Data": "/usr/src/freeswitch/sounds/basic/welcome.wav", "Application-Response": "FILE PLAYED", "Application-UUID": "ca0df0a8-e5be-11e3-a83e-551ac922334e"

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URI dialplan setup
by Mark Campbell-Smith 30 May '14

30 May '14

Hi Voip provider Ippi allow calls to/from skype users. For me, this will work quite well in my home pabx, and I do not want to implement the SipThreeSkype solution. I have created the gateway and successfully registered to Ippi's sip server at ippi.fr. Ippi requires that to dial a skype user, the following URI format must be dialed: my-skype-address(a)skype.ippi.com. (this successfully calls my skype address using a softphone) I am trying to create a dialplan that will allow me to call a skype user through the Ippi gateway. This is my gateway setup (notice ippi uses ippi.fr): <include> <gateway name="ippi.fr"> <param name="username" value="username"/> <param name="password" value="password"/> <param name="proxy" value="ippi.fr"/> <param name="realm" value="ippi.fr"/> <param name="auth-username" value="username"/> <param name="register" value="true"/> </gateway> </include> I have the following in the dialplan: <extension name="ippi-skype"> <condition field="destination_number" expression="^5000$"> <action application="set" data="call_timeout=90"/> <action application="set" data="hangup_after_bridge=true"/> <action application="info"/> <action application="bridge" data="sofia/gateway/ ippi.fr/my-skype-address(a)skype.ippi.com"/> </condition> </extension> Now I realize that according to the wiki, I should not put sofia/gateway/ ippi.fr/my-skype-address(a)skype.ippi.com (ie the @skype.ippi.com), but I need a way to dial out through the ippi gateway (ippi.fr), but to a user in domain skype.ippi.com. I did try the two following options: <action application="set" data="sip_invite_to_uri= my-skype-address(a)skype.ippi.com"/> <action application="export" data="sip_invite_to_uri= my-skype-address(a)skype.ippi.com"/> Wireshark gives me the following for a successful call using Jitsi softphone: Session Initiation Protocol Request-Line: INVITE sip:my-skype-address@skype.ippi.com SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Call-ID: 891b23da468c6fdf05434ac89f123cf1@0:0:0:0:0:0:0:0 CSeq: 1 INVITE Sequence Number: 1 Method: INVITE From: "username" <sip:username@ippi.fr>;tag=35bcc007 SIP Display info: "username" SIP from address: sip:username@ippi.fr SIP tag: 35bcc007 To: <sip:my-skype-address@skype.ippi.com> SIP to address: sip:my-skype-address@skype.ippi.com Via: SIP/2.0/TCP 192.168.1.6:62110 ;branch=z9hG4bK-373232-f15d6a66ae2d195ba99f1d308e43e61d Transport: TCP Sent-by Address: 192.168.1.6 Sent-by port: 62110 Branch: z9hG4bK-373232-f15d6a66ae2d195ba99f1d308e43e61d Max-Forwards: 70 Contact: "username" <sip:username@192.168.1.6:62110 ;transport=tcp;registering_acc=ippi_fr> Contact Binding: "username" <sip:username@192.168.1.6:62110 ;transport=tcp;registering_acc=ippi_fr> User-Agent: Jitsi2.5.5137Windows 7 Content-Type: application/sdp Content-Length: 910 What do I use in the bridge statement so that it calls my-skype-address(a)skype.ippi.com, using gateway ippi.fr? Suggestions appreciated!

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NetBSD/OpenBSD/DragonFlyBSD Ports
by Brian West 30 May '14

30 May '14

FreeSWITCHers, Looking for interested parties to help build and maintain ports of FreeSWITCh on these platforms, I'm already working with someone on FreeBSD to get those updated. Please email me off list we can schedule a call to discuss this in a community project as we do with the FS-Docs team, We'll just have a FS-Portability team community meeting ;) Again you don't have to be a wizard with C or autotools, just proficient on the target platform with a interested in helping spread FreeSWITCH. 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)

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