April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
by technologyinspired 21 Apr '09

21 Apr '09

Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Thanks, Regards, Vin

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Experience with libpri?
by Peter P GMX 21 Apr '09

21 Apr '09

In the past I had a number of problems with PRI and OpenZAP (timer problems). I saw in this mailing list and in the wiki that FS supports libpri since a while. Can you share your experience with libpri? Is it considered to be stable in production? Are ther any issues I have to take care about? I am thinking of setting up a FS system with Sangoma A108 8 Ports PRI card and therefore I need a really stable PRI implementation. Best regards Peter

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Errors while installing FreeSwitch 1.0.4pre4.
by technologyinspired 21 Apr '09

21 Apr '09

I have found erros while installing FreeSwitch 1.0.4pre4. I am using Toshiba Laptop with FC10. here is the partial output of "make" command: ... ... Making servers/slapd/backends.c Add config ... Add ldif ... Making servers/slapd/overlays/statover.c Please run "make depend" to build dependencies Making all in /usr/src/freeswitch-1.0.4pre4/libs/openldap-2.4.11 Entering subdirectory include Making ldap_config.h Entering subdirectory libraries Making all in /usr/src/freeswitch-1.0.4pre4/libs/openldap-2.4.11/libraries Entering subdirectory liblutil getpeereid.c: In function ‘lutil_getpeereid’: getpeereid.c:65: error: storage size of ‘peercred’ isn’t known make[8]: *** [getpeereid.o] Error 1 make[7]: *** [all-common] Error 1 make[6]: *** [all-common] Error 1 make[5]: *** [/usr/src/freeswitch-1.0.4pre4/libs/openldap-2.4.11/libraries/libldap_r/ libldap_r.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root@localhost freeswitch-1.0.4pre4] Even though it is showing that FreeSwitch has been successfully buit and run make install. at the end it gives 2 errors. So is the "make" process complete or there is some bug? Regards, Vin

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ekiga and freeswitch
by e schmidbauer 21 Apr '09

21 Apr '09

I am running FreeSWITCH Version 1.0.3 on a centos 5.2 x64 server and ekiga 3.2.0 on a client computer. I am able to register the ekiga client with the freeswitch server but when I dial an extension to join a conference I get the following error message: [ERR] switch_core_io.c:327 switch_core_session_read_frame() Codec RAW Signed Linear (16 bit) decoder error! I have the celt codec loaded on both the server and the client computer. This is the full output when dialing an extension: 2009-04-12 12:56:34 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing joseph schmidbauer->4800 in context default 2009-04-12 12:56:34 [INFO] mod_sofia.c:1301 sofia_receive_message() Asked to send early media by sofia/internal/joe(a)192.168.1.125 2009-04-12 12:56:34 [INFO] mod_sofia.c:1342 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1239529036 1239529037 IN IP4 192.168.1.125 s=FreeSWITCH c=IN IP4 192.168.1.125 t=0 0 m=audio 26358 RTP/AVP 116 101 a=rtpmap:116 CELT/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:10 a=sendrecv 2009-04-12 12:56:34 [NOTICE] mod_sofia.c:1345 sofia_receive_message() Pre-Answer sofia/internal/joe(a)192.168.1.125! 2009-04-12 12:56:34 [NOTICE] mod_conference.c:1934 conference_loop_output() Channel [sofia/internal/joe(a)192.168.1.125] has been answered warning: decode error [ERR] switch_core_io.c:327 switch_core_session_read_frame() Codec RAW Signed Linear (16 bit) decoder error! [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/internal/joe(a)192.168.1.125 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-12 13:05:14 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 4 (sofia/internal/joe(a)192.168.1.125) Ended 2009-04-12 13:05:14 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/joe(a)192.168.1.125 [CS_HANGUP] Not sure if this is a bug in the program or just in my setup. I've tried using the svn version of freeswitch (as of yesterday) and i got the exact same error. Any input would be appreciated. Thanks!

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fifo taking > 5 seconds to bridge calls
by Matthew Fong 21 Apr '09

21 Apr '09

I tried using fifo in an environment with about 9 agents last week, but ran into some issues, that I'm trying to piece together. The system is setup on a new ubuntu 64-bit machine and it should be plenty fast to handle this load. The delay does not occur when testing with a single agent...so it's hard for me to replicate on my vmware-based development machine, but thought I'd toss this out to the community to see if anyone has any suggestions I'm using fifo in the reverse sense, in that my "agents" are put into a fifo, and an outbound call (a "contact") take an agent out of the fifo once the call is answered. I did this because it is an outbound call center, not an inbound one, and in reality I have agents waiting for a contact...not callers waiting for an agent. Anyway, on about all my calls the logs indicate (and the agents I'm working with confirm), it appears to be taking > 5 seconds to actually bridge a call once the outbound "contact" channel is answered by FS and the channel takes an agent out of the fifo. I've attached a snipet of the fs.log and the dialplan below. But from timestamps it appears the delay is caused somewhere between these EXECUTE commands EXECUTE sofia/external/1XXX4951027 set_user(default@) EXECUTE sofia/external/1XXX4951027 db(insert/-spymap/1800XXX8234/aa763964-2933-11de-bbea-318f8f194b60) EXECUTE sofia/external/1XXX4951027 db(insert/-last_dial/1800XXX8234/1990) EXECUTE sofia/external/1XXX4951027 db(insert/-last_dial/global/aa763964-2933-11de-bbea-318f8f194b60) EXECUTE sofia/external/1XXX4951027 answer() EXECUTE sofia/external/1XXX4951027 set(fifo_caller_consumer_import=hh_stomp,hh_user) when I looked at this..nothing pops out as out of the ordinary, that should cause >5 seconds delay. Your thoughts please .. Thanks. --matt Full Log: 2009-04-14 16:35:18 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/external/1XXX4951027 SOFIA EXECUTE 2009-04-14 16:35:18 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/external/1XXX4951027 Standard EXECUTE EXECUTE sofia/external/1XXX4951027 set(open=true) 2009-04-14 16:35:18 [DEBUG] mod_dptools.c:748 set_function() sofia/external/1XXX4951027 SET [open]=[true] EXECUTE sofia/external/1XXX4951027 set(use_profile=default) 2009-04-14 16:35:18 [DEBUG] mod_dptools.c:748 set_function() sofia/external/1XXX4951027 SET [use_profile]=[default] EXECUTE sofia/external/1XXX4951027 set_user(default@) EXECUTE sofia/external/1XXX4951027 db(insert/-spymap/1800XXX8234/aa763964-2933-11de-bbea-318f8f194b60) EXECUTE sofia/external/1XXX4951027 db(insert/-last_dial/1800XXX8234/1990) EXECUTE sofia/external/1XXX4951027 db(insert/-last_dial/global/aa763964-2933-11de-bbea-318f8f194b60) EXECUTE sofia/external/1XXX4951027 answer() EXECUTE sofia/external/1XXX4951027 set(fifo_caller_consumer_import=hh_stomp,hh_user) 2009-04-14 16:35:42 [DEBUG] mod_dptools.c:748 set_function() sofia/external/1XXX4951027 SET [fifo_caller_consumer_import]=[hh_stomp,hh_user] EXECUTE sofia/external/1XXX4951027 set(fifo_consumer_exit_key=5) 2009-04-14 16:35:42 [DEBUG] mod_dptools.c:748 set_function() sofia/external/1XXX4951027 SET [fifo_consumer_exit_key]=[5] EXECUTE sofia/external/1XXX4951027 fifo(2 out wait undef undef) 2009-04-14 16:35:42 [DEBUG] switch_channel.c:182 switch_channel_audio_sync() sofia/external/1XXX4951027 receive message [AUDIO_SYNC] 2009-04-14 16:35:43 [DEBUG] switch_ivr_bridge.c:911 switch_ivr_multi_threaded_bridge() sofia/external/1XXX4951027 receive message [BRIDGE] Dialplan that Inbound "Contact" Calls are placed in <extension name="1990"> <condition field="destination_number" expression="^1990$"> <action application="answer"/> <action application="set" data="fifo_caller_consumer_import=hh_stomp,hh_user"/> <action application="set" data="fifo_consumer_exit_key=5"/> <action application="fifo" data="${ag_qcall} out wait undef ${ag_dnc_msg}"/> </condition> </extension> Dialplan for "my" Agent Calls: <extension name="1992"> <condition field="destination_number" expression="^1992$"> <action application="playback" data="hh/hh-doorbell.wav"/> <action application="fifo" data="${ag_qcall} in currency/dollars.wav $${hold_music}"/> </condition> </extension>

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Re: [Freeswitch-users] Configure FS using Flowroute.com
by Chris Fowler 21 Apr '09

21 Apr '09

You've called your gateway "flowroute" vs. mine which is "sip.flowroute.com"; you need to amend the dialplan to reference "flowroute"; then you should be set. >>>>> flowroute gateway sip:*******@sip.flowroute.com <action application="bridge" data="sofia/gateway/flowroute/${default_provider_username}#$1"/> C. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alex Harris Sent: Monday, April 20, 2009 20:22 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com The vars.xml has been modified with the following: <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=*******"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=********"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=sip.flowroute.com"/>  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> the 01_example.xml has been modified with the code you provided The Status output is: Name Type Data State ============================================================================ =============================== internal profile sip:mod_sofia@**.***.219.221:5060 RUNNING (0) external profile sip:mod_sofia@**.***.219.221:5080 RUNNING (0) flowroute gateway sip:*******@sip.flowroute.com REGED **.***.219.221 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[2002:63be:dbdd::63be:dbdd]:5060 RUNNING (0) default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED ============================================================================ ============================== When I try to dial out this is what I am getting: freeswitch@S4BS> 2009-04-20 20:19:45 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/internal/1001(a)**.***.219.221 [7bbb17eb-e953-1743-84b5-d4b0ae651332] 2009-04-20 20:19:45 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Alex->15184282539 in context default 2009-04-20 20:19:45 [ERR] mod_sofia.c:2411 sofia_outgoing_channel() Invalid Gateway 2009-04-20 20:19:45 [NOTICE] mod_sofia.c:2624 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-04-20 20:19:45 [ERR] switch_ivr_originate.c:1459 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-04-20 20:19:45 [INFO] mod_dptools.c:2036 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT 2009-04-20 20:19:45 [NOTICE] mod_dptools.c:2068 audio_bridge_function() Hangup s ofia/internal/1001(a)**.***.219.221 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1018 switch_core_session_thre ad() Session 3 (sofia/internal/1001(a)**.***.219.221) Ended 2009-04-20 20:19:45 [NOTICE] switch_core_session.c:1020 switch_core_session_thre ad() Close Channel sofia/internal/1001(a)**.***.219.221 [CS_DONE] Thank you for the help. ~Alex -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Chris Fowler Sent: Monday, April 20, 2009 7:09 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com Are you either restarting FS or issuing the reloadxml command (press F6 on the console) after making these changes? Did you modify vars.xml per my last note? Also, check out http://wiki.freeswitch.org/wiki/Getting_Started_Guide - it's worth investing the time to understand how FS parses the various config files. Look in /usr/local/freeswitch/log at freeswitch.xml.fsxml - this file contains the entire free switch configuration since last as parsed by the application. Seems you're not hitting the issue I was with FlowRoute - but as the error indicates you're trying to route a call out from a gateway that does not exist. What's the output of "sofia status" (F5 on the console)? It should show: sip.flowroute.com gateway sip:nnnnnnn@sip.flowroute.com REGED Chris. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alex Harris Sent: Monday, April 20, 2009 17:05 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Configure FS using Flowroute.com I modified the 01_example.com.xml with the code below and I am still getting "Invalid Gateway". I tried creating a file 01_flowroute.com.xml and placed the code below and had the same unfortunate result. On flowroute site I have placed my IP in the outbound allowed IPs and also inbound routes. Is there something else that I should do to try to get things to work? Sorry, for asking for so much help. ~Alex _______________________________________________ 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 _______________________________________________ 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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How to use variables in console/event socket
by Szymon Olko 21 Apr '09

21 Apr '09

Hello, I don't know how to use variables in console. I try to call local user with following commands. Command: originate sofia/internal/1001%${domain} 9999 I get: Cannot locate registered user 1001@%{domain} Command: originate sofia/internal/1001@${domain} 9999 I get: Cannot locate registered user 1001@@{domain} Command: originate sofia/internal/1001%\${domain} 9999 I get: Cannot locate registered user 1001@\%{domain} Substitution of $ sign is not understood for me. Is it possible to use variables in that channels? Variable is defined: freeswitch@vertux> global_getvar domain API CALL [global_getvar(domain)] output: 192.168.77.248 Szymon O.

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NAT between FS and remote SIP phone
by Fred-145 21 Apr '09

21 Apr '09

Hello I'd like to know how to set things up when using the following scenario: - a VoIP gateway on the same LAN as the Freeswitch to handle incoming calls from a POTS line - a remote SIP phone somewhere on the Net - the FS server and the remote SIP phone are both behind a NAT router - the remote SIP user either doesn't have the computer skills to map ports on his NAT router, or doesn't have access to it (eg. staying in a hotel or connecting to FS from a wifi connection @ Starbucks) The questions I have: 1. What ports need to be mapped on each router? 2. If I understood it correctly, UPnP is a technology that can open ports dynamically. Are there ways to tell if a router supports UPnP, are there other ways to have a remote SIP phone work right out of the box, or are there cases where mapping ports manually is the only way to get SIP/RTP to work? 3. When a call comes from the POTS line and meant for the remote SIP extension... do RTP packets flow directly from the Linksys VoIP gateway to the remote SIP phone, or do they go through the FS server? Thank you for any hint. -- View this message in context: http://www.nabble.com/NAT-between-FS-and-remote-SIP-phone-tp23136640p231366… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Quick question on how XML files are handled
by Fred-145 21 Apr '09

21 Apr '09

Hello I'm reading through the articles on the wiki, and have a basic question about how Freeswitch handles the XML files: I noticed that conf/sip_profiles contains both internal.xml, internal/, and internal/example.xml: Does it mean that Freeswitch merges everything it finds in this location (in this case, stuff in internal.xml + internal/*.xml), or does it depend on what is included in freeswitch.xml? Thank you. -- View this message in context: http://www.nabble.com/Quick-question-on-how-XML-files-are-handled-tp2313626… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Best G729 replacement
by Nicolas Brenner 21 Apr '09

21 Apr '09

Hi, I might be in a position (finally) to ask/suggest one of my voip providers to use an alternative codec to G729. I wanted to know what would be the best replacement for it. Thanks again everybody for your time and info. Regards, Nicolas

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