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Clustering FreeSWITCH
by Marc Lewis 29 Oct '08

29 Oct '08

I am in the process of making my FreeSWITCH installation highly available and I'm running into a couple of snags that was hoping that someone may have some insight on. First, the setup as it is now. There are two installations of FS on two different servers, lets call them fs1 and fs2. They each pull their configurations, dialplan, directory and post CDR's all using mod_curl from a central web server. That part works great. Calls into and out of FS go through an OpenSER proxy set up using carrierroute. That part also works great for outbound calls to the PSTN. Inbound calls also come in through this OpenSER proxy and get routed to the primary switch fs1. That also works perfectly as long as its going to fs1. fs1 and fs2 are both setup to use an ODBC connection to store registrations. This is pointed to a MySQL database made highly available using the RedHat Cluster Suite on a shared fibre channel partition. fs1 and fs2 both share the same database. Voicemail storage on fs1 is directly mounted on a GFS2 partition, fs2 is mounting the shared storage from a different server via NFS for no single point of failure. For the phones, I have them setup to use SRV records and have fs1 at priority 10 and fs2 at priority 20 for acme.domain.com. I've tested this and phones register to the correct server and the sip_registration table shows either fs1 or fs2 as the hostname as I would expect. Here is the problem. If user 100(a)acme.domain.com registers on fs2 and a call comes in from the OpenSER proxy to fs1, bridging the call to /sofia/internal/100%acme.domain.com from fs1 doesn't ring the phone. Is there a difference between 'sofia/internal/100%acme.domain.com' and 'user/100(a)acme.domain.com'? Calls out from either fs1 or fs2 routed to the proxy work fine, its just calls coming in from the proxy. If the call doesn't go to the switch the user is registered on, the user's phone doesn't ring. It still goes to voicemail, etc., so that part works. Is there a better way to cluster FreeSWITCH than DNS SRV records and a shared state database? Also, as a side note to Anthony, Brian, et al, if this is the best way, I'll be happy to write up a wiki page on how I have this setup with a lot more detail than this. I was not able to find much in the way of highly available configurations or cluster configurations, so I put together this system using information cobbled from the wiki, mailing list messages and lurking on IRC. Thanks. - Marc -- Marc Lewis Avvatel Corporation

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Help! No output to CLI with console_log() from script
by mszlazak＠aol.com 29 Oct '08

29 Oct '08

I'm trying to get output to my CLI in Windows XP when running this script by dialing 8337 but the output "Hello World!" doesn't show up. The rest works fine. What's wrong with console_log("Hello World!\n") ? Where has the output gone? console_log("Hello World!\n"); var languageCode = "en"; var soundDir = "sound/"; function playFile(fileName, callBack, callBackArgs) { ??? session.streamFile(soundDir + languageCode +? "/" + fileName); } session.answer(); playFile("HelloWorld.wav"); exit();

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freeswitch as a stateful SIP proxy or back2back user agent?
by Ryan McDougall 29 Oct '08

29 Oct '08

Apologies if this has been answered somewhere already, but does freeswitch expose an API that would make it appropriate as a SIP proxy? Any advice you could share would be greatly appreciated. Cheers,

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Cepstral 5.1 no sound
by Peter P GMX 29 Oct '08

29 Oct '08

Hello, I receive the following message during CS_INIT *Failed to load library libceplang_de.so due to: /opt/swift/lib/libceplang_de.so: undefined symbol: cst_rx_int* Later however, FS at least tries to speak: 2008-10-28 23:40:05 [NOTICE] mod_dptools.c:605 answer_function() Channel [sofia/internal/1005(a)siplocal.safecomm.ch] has been answered 2008-10-28 23:40:05 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/internal/1005(a)siplocal.safecomm.ch Execute speak(cepstral|Katrin|Sie werden jetzt ins Festnetz weiterverbunden) 2008-10-28 23:40:05 [DEBUG] switch_ivr_play_say.c:1911 switch_ivr_speak_text() OPEN TTS cepstral 2008-10-28 23:40:05 [DEBUG] switch_ivr_play_say.c:1920 switch_ivr_speak_text() Raw Codec Activated 2008-10-28 23:40:05 [DEBUG] switch_ivr_play_say.c:1636 switch_ivr_speak_text_handle() Speaking text: Sie werden jetzt ins Festnetz weiterverbunden 2008-10-28 23:40:05 [DEBUG] sofia.c:2259 sofia_handle_sip_i_state() Channel sofia/internal/1005(a)siplocal.safecomm.ch entering state [completed] 2008-10-28 23:40:05 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Send signal sofia/internal/1005(a)siplocal.safecomm.ch [BREAK] but I can hear no sound. Is it because I have installed Cepstral 5.1? Do I need 4.2? P.S.: Environment var SWIFT_HOME is set. Best regards Peter

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System requirements
by Jonas Gauffin 29 Oct '08

29 Oct '08

Hello What kind of system requirements do FS have to handle about 500 simultaneous calls? The specsheet in the wiki have Minimum/Recommended System Requirements, but they just talk about memory and disk space (and doesnt mention how many calls the recomendations are for). Thanks, Jonas

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FreeSWITCH as pure SIP proxy
by Arturo Díaz Almagro 28 Oct '08

28 Oct '08

Thanks for your answers. In fact I am using Kamailio as SIP proxy, but some additional functionality is needed that Kamailio cannot afford. I need a B2BUA in the middle of the signaling between phones and Kamailio to get track of dialogs and to interfere on them when neccessary. For that reason I need FS to receive REGISTER messages and to forward them to Kamailio just adding a Via record. From your answers I conclude that FS cannot forward those REGISTER methods, don't I? Again, thanks for your answers. I will try to investigate a lot more about FS and its capabilities. Regards. -- Arturo Díaz Contact me on FWD: 870436 Skype: arturo.diaz.almagro

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Take uuid out of conference and bridge
by Birgit Arkesteijn 28 Oct '08

28 Oct '08

Hi, In javascript I'm trying to take 'memberA' out of a conference and bridge this person to a customer services number. I no longer have the session object of memberA, so I find the uuid via an apiExecute() on conference list. I then create a new session object with the customer services endpoint, and try to uuid_bridge the uuid of memberA with the uuid of the new session object. The customer service endpoint ring successfully, but when I try to bridge both endpoints hangup with cause DESTINATION_OUT_OF_ORDER. Does anyone know how to do this successfully? Here is the function involved: function bridge_to_customer_services(conf_name, member_id, uuid, customer_service_url) { if (member_id && uuid) { var sSession = new Session("{ignore_early_media=true,originate_timeout=10}" + customer_service_url); var ready = sSession.ready(); if (ready == true) { // The customer services call has been answered var argument = conf_name + " kick " + member_id; var res = apiExecute("conference", argument); log("after apiExecute conference: " + res); argument = uuid + " " + sSession.uuid; res = apiExecute("uuid_bridge", argument); log("after apiExecute uuid_bridge: " + res); } } } BTW, log() is just a helper function that wraps around console_log(). Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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DTMF Star Event Inconsistent
by Klaus Teller 28 Oct '08

28 Oct '08

Hi, I'm calling a registered soft phone (ext. 1003) via the event socket interface. That is, on one side i have some Java code connecting to the Freeswitch event socket interface and placing calls and on the other hand i have the soft phone registered to Freeswitch and awaiting for calls. Now, when i get a call on the soft phone, i press a sequence of DTMF digits. The sequence of DTMF digits is intended to be read by the Java code via the socket interface. Most things run pretty smoothly: i can place calls, i can send DTMFs, i can receive them on the other hand. The inconsistent behavior i'm seeing is following. For DTMF-0 to DTMF-9, and DTMF-#, i receive two events via the socket interface. The first one is in CS_EXCHANGE_MEDIA state and the second is in CS_EXECUTE state. Yet for DTMF-* i receive inconsistent number of events: sometimes only one single event in state CS_EXCHANGE_MEDIA sometimes two events as in the case of other DTMF digits. It seems there is a pattern in this "inconsistency". The odd DTMF-* (first, third, fifth, etc.) generate only one CS_EXCHANGE_MEDIA event while the even (second, fourth, sixth, etc.) generate both events. Can somebody help me understand what's going on? Thanks, Klaus. -- "Feel free" - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: http://www.gmx.net/de/go/promail

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Maximum Calls In Progress
by jocke eriksson 28 Oct '08

28 Oct '08

Hello FS! I have a question about un earlier thread, the one where FS returns a 503 response when ongoing calls reaches 240-250. I think Ron had the same problem, so my question is, was the problem resolved? And if so, what did he do?

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Javascript Quickstart page not as "quick" as it could be
by mszlazak＠aol.com 28 Oct '08

28 Oct '08

I got FreeSwitch running today and connected a softphone to it. Yippee! But ran into some snags trying out Javascript within a dialplan. My FreeSwitch is running on Windows XP and from the same machine I use an X-Lite softphone to call and play around while learning how Javascript and FreeSwitch work together. The first snag was the location of the default dialplan file given in the "Quickstart" (http://wiki.freeswitch.org/wiki/Javascript_QuickStart) It is "conf\dialplan\default.xml" on my system and not as the "Quickstart says in "conf/default_context" since "default_context" does not exist.? ? Next, trying the JavaScript Quickstart HelloWorld.js example, I get the following errors which indicate a different location is needed for the .wav file from what the "Quickstart" says and this also requires two more new folders (sound & en) nested inside folder callie. Now, either I did something wrong or the instructions need more updating. ???? ? 2008-10-27 22:34:16 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing FreeSwitch->8337 in context default 2008-10-27 22:34:16 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000(a)10.0.0.2] has been answered 2008-10-27 22:34:16 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [C:\Program Files\FreeSWITCH/sounds/en/us/callie\sound/en/HelloWorld .wav] [System error : The system cannot find the path specified. ] 2008-10-27 22:34:16 [ERR] HelloWorld.js:6 mod_spidermonkey()? TypeError: callBack is not a function 2008-10-27 22:34:16 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/internal/1000(a)10.0.0.2 [CS_EXECUTE] [NORMAL _CLEARING] 2008-10-27 22:34:16 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 3 (sofia/internal/1000(a)10.0.0.2) Ended 2008-10-27 22:34:16 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.0.0.2 [CS_HANGUP] I suppose I could have not used variables "languageCode" and "soundDir" but I decided to create the two new folders and placed the .wav file in folder "en" (C:\Program Files\FreeSWITCH/sounds/en/us/callie\sound/en/HelloWorld.wav). Also, I removed the arguments 'callBack' and 'callBackArgs' from 'session.streamFile(soundDir + languageCode + "/" + fileName, callBack, callBackArgs);'. Otherwise, I would get the following error: 2008-10-27 22:58:08 [ERR] HelloWorld.js:6 mod_spidermonkey()? TypeError: callBack is not a function This gives me an updated HelloWorld.js file that looks like this: var languageCode = "en"; var soundDir = "sound/"; function playFile(fileName, callBack, callBackArgs) { ?? ?session.streamFile(soundDir + languageCode +? "/" + fileName); } session.answer(); playFile("HelloWorld.wav"); exit(); The command line interface gave the following and I finally heard my HelloWorld.wav file :-) 2008-10-27 23:02:59 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/1000(a)10.0.0.2 [056e90b9-e49a-d544-8076-573fbeb50 9f3] 2008-10-27 23:02:59 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing FreeSwitch->8337 in context default 2008-10-27 23:02:59 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000(a)10.0.0.2] has been answered 2008-10-27 23:03:10 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/internal/1000(a)10.0.0.2 [CS_EXECUTE] [NORMAL _CLEARING] 2008-10-27 23:03:10 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 9 (sofia/internal/1000(a)10.0.0.2) Ended 2008-10-27 23:03:10 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.0.0.2 [CS_HANGUP] However, the sound has was quite static filled. What's the cause of all that static and how do I fix (remove) it? Thanks. Mark.

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