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Can't get freeswitch to send BYE packet on hangup
by Philipp Schrader 21 Jun '10

21 Jun '10

Dear All, Loving freeswitch so far. There's one thing I need help with though. I can't seem to get freeswitch to send BYE packets to my originating call when the hangup happens. At first I tried this in Lua, but the symptoms are still the same with a barebone dialplan: <extension name="bridge_call_test"> <condition field="destination_number" expression="^103"> <action application="answer"/> <action application="sleep" data="5000"/> <action application="hangup"/> </condition> </extension> I also turned siptrace on with the commands: sofia profile internal siptrace on sofia profile external siptrace on All I can get though is a SIP/2.0 200 OK at the beginning of the call, but no BYE at the end: Does anyone have an idea as to what I am missing? 2010-06-21 13:10:20.600232 [INFO] switch_cpp.cpp:1142 username: XXXX, password: XXXXXXXX 2010-06-21 13:10:20.603790 [NOTICE] switch_channel.c:669 New Channel sofia/internal/XXXX@aaaaa [6a2c61cc-4f44-4a33-b6fc- af9728c2f1d2] 2010-06-21 13:10:20.607511 [INFO] mod_dialplan_xml.c:418 Processing XXXX->103 in context default 2010-06-21 13:10:20.618949 [NOTICE] mod_dptools.c:719 Channel [sofia/internal/XXXX@aaaaa] has been answered send 1228 bytes to udp/[192.168.202.67]:5060 at 17:10:20.619691: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.202.67:5060 ;rport=5060;branch=z9hG4bKPjf2d4bf79-38fb-45fe-82bf-13b070750f90 From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff To: <sip:103@aaaaa>;tag=8rcH55QDQpmDD Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a CSeq: 24285 INVITE Contact: <sip:103@192.168.202.31:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 251 Remote-Party-ID: "103" <sip:103@192.168.202.31 <sip%3A103(a)192.168.202.31> >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1277110358 1277110359 IN IP4 192.168.202.31 s=FreeSWITCH c=IN IP4 192.168.202.31 t=0 0 m=audio 29862 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 386 bytes from udp/[192.168.202.67]:5060 at 17:10:20.628813: ------------------------------------------------------------------------ ACK sip:103@192.168.202.31:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.202.67:5060 ;rport;branch=z9hG4bKPj65f1608c-62e2-4ca8-973b-8d8daf8a9a6b Max-Forwards: 70 From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff To: sip:103@aaaaa;tag=8rcH55QDQpmDD Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a CSeq: 24285 ACK Content-Length: 0 ------------------------------------------------------------------------ 2010-06-21 13:10:25.618976 [NOTICE] mod_dptools.c:705 Hangup sofia/internal/XXXX@aaaaa [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1182 Session 2 (sofia/internal/XXXX@aaaaa) Ended 2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/XXXX@aaaaa [CS_DESTROY]

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Re: [Freeswitch-users] voicemail wav filename
by Madovsky 21 Jun '10

21 Jun '10

I'd like to change the name of the voicemail attached. but can't find the right conf file.. anyone ? Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, June 21, 2010 12:47 PM Subject: voicemail wav filename Hi, where can I customize the wav filename sent from voicemail ? Thanks F

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span not defined error
by Neil Patel 21 Jun '10

21 Jun '10

Hi All, I am trying to dial out over my PRI, and am getting this error: 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED! 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION] This is my openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 This is my openzap.conf.xml: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </boost_spans> <analog_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </analog_spans> </configuration> And here is the lua code I'm using to dial out: sessiondata = "OpenZAP/smg_prid/<num>" new_session = freeswitch.Session(sessiondata) What am I missing here? Thanks, Neil

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voicemail wav filename
by Madovsky 21 Jun '10

21 Jun '10

Hi, where can I customize the wav filename sent from voicemail ? Thanks F

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session variable question
by Stephen Cattaneo 21 Jun '10

21 Jun '10

Are the following the same thing? i just want to make sure because i've seen both in example scripts. session.setVariable("hangup_after_bridge",true); session.execute("set","hangup_after_bridge=true"); Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print

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t.38 in newer fs versions - t38-passthru
by Christian Löschenkohl 21 Jun '10

21 Jun '10

hello has anybody tried the config parameter t38-passthru in a sip profile? our fs server core dumped today after i did set the option an did a reladxml + a profile rescan does this option bring back the old behavior that the proxy mode provided? the actual behavior (actual git from 17.06.2010) is not useable. t.38 isn't passed through without modification and fax sending fails almost every time. in e.g. version 1.0.6 t.38 in proxy mode worked 100% reliable br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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mod_limit / db / hash changes - If you update action is required
by Rupa Schomaker 21 Jun '10

21 Jun '10

If you've been paying attention to the conf call or irc you'll know this change was coming. I've committed a significant change to how limits are managed in FreeSWITCH. This change is NOT backwards compatible. It requires one to: 0) git pull (of course) 1) rerun ./configure to generate new Makefiles for new modules 2) edit modules.conf, remove mod_limit and add mod_db and mod_hash 3) modify conf/autoload_configs/modules.conf.xml, remove mod_limit and add mod_db and mod_hash 4) create/edit conf/autoload_configs/db.conf.xml if using ODBC In your dialplan, anywhere you are using limit, you need to add db to the data line. Anywhere you are using limit_hash you need to remove _hash from application and add hash to the front of the data line. Same pattern applies to any API usage of limit you may be using. Read the wiki at: http://wiki.freeswitch.org/wiki/Limit for documentation on limit. Ok, why the change? limit has been moved into core with support for pluggable backend limit providers. This allows one to develop limit backends relatively easily and provides a more structured interface to these different limit backends. Expect to see backends provided by more than just db and hash. It also always bothered me that db and hash were part of the mod_limit module. They are not standalone and provide functionality specific to themselves. I've updated the wiki with documentation. I'll continue to update the wiki as necessary, feel free to do so yourself if there is something unclear or missing. -- -Rupa

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Start/stop recording controls and multiple recordings of a single conference.
by M. Ranganathan 20 Jun '10

20 Jun '10

Hello, I have a couple of questions about the behavior of conference recording: 1. In Looking at the semantics of norecord as stated in, http://wiki.freeswitch.org/wiki/Mod_conference it is not clear to me whether or not it is possible to start/stop/restart recording of an ongoing conference. Any clarifications would be appreciated. 2. Can you have multiple recordings for a given conference? i.e. conference foo record /tmp/recording1.wav and conference foo record /tmp/recording2.wav at the same time for a single ongoing conference. Thank you in advance for any help. Regards, Ranga -- M. Ranganathan

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check if transcoding is needed
by Pete Kay 20 Jun '10

20 Jun '10

Hi, Is there anyway to detect/check if codec transcoding is needed? What I would like to do is to set "bypass_media=true" if no transcoding is needed. Any suggestion on how / when should bypass_media be turned on/off? thanks, P

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question about NibbleBill
by Madovsky 20 Jun '10

20 Jun '10

hi there, I'm trying to create user who can call for free and other not. I set nobal_action in nibblebill conf to an extension in my default dialplan but if the user is at $0 for special case I would authorize the call for free. is it possible ? Thanks F

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