December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

OpenSIPS Load Balancer and FreeSwitch issue
by Peter Spinato 19 Dec '11

19 Dec '11

All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter

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destination_number cleanup
by Elliott Vogel 19 Dec '11

19 Dec '11

I was wondering if anyone has a regex expression that works to return just digest? I have some clients sending requests formatted to +1 (555) 555-5555, 555-555-5555, 555.555.5555 which aren't be processed by our dial plan because we are expecting all numbers (5555555555)

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Where are table definitions of database for freewsitch (latest GIT version)
by king2kin 19 Dec '11

19 Dec '11

Hi there, On Dec. 18, 2011, I dowloaded the latest GIT version of freeswitch source codes by git clone git://git.freeswitch.org/freeswitch.git Then, I compiled the project solution with vs2008, and run FreeSwitch.exe from command-line on windows 2003. I got a bunch of errors complaining about SQL errors, for example, { switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] } Unlike old version, the new version on windows system seems to require setting up extra database tables according to log file [freeswitch.log].The log file gave basic definitions (see below) of these SQL tables, however, primary key and/or constraints are NOT provided. Could anyone tell me where to find the full definition of these SQL tables that are required by the latest GIT version of freeswitch? Thanks in adavance! ==== table definitions gave in freeswitch.log ========================= -- -- table [sip_registrations] -- CREATE TABLE sip_registrations ( call_id VARCHAR(255), sip_user VARCHAR(255), sip_host VARCHAR(255), presence_hosts VARCHAR(255), contact VARCHAR(1024), status VARCHAR(255), rpid VARCHAR(255), expires INTEGER, user_agent VARCHAR(255), server_user VARCHAR(255), server_host VARCHAR(255), profile_name VARCHAR(255), hostname VARCHAR(255), network_ip VARCHAR(255), network_port VARCHAR(6), sip_username VARCHAR(255), sip_realm VARCHAR(255), mwi_user VARCHAR(255), mwi_host VARCHAR(255), orig_server_host VARCHAR(255), orig_hostname VARCHAR(255) ); -- -- table [sip_subscriptions] -- CREATE TABLE sip_subscriptions ( proto VARCHAR(255), sip_user VARCHAR(255), sip_host VARCHAR(255), sub_to_user VARCHAR(255), sub_to_host VARCHAR(255), presence_hosts VARCHAR(255), event VARCHAR(255), contact VARCHAR(1024), call_id VARCHAR(255), full_from VARCHAR(255), full_via VARCHAR(255), expires INTEGER, user_agent VARCHAR(255), accept VARCHAR(255), profile_name VARCHAR(255), hostname VARCHAR(255), network_port VARCHAR(6), network_ip VARCHAR(255), version INTEGER DEFAULT 0 NOT NULL, orig_proto VARCHAR(255), full_to VARCHAR(255) ); -- -- table [sip_dialogs] -- CREATE TABLE sip_dialogs ( call_id VARCHAR(255), uuid VARCHAR(255), sip_to_user VARCHAR(255), sip_to_host VARCHAR(255), sip_from_user VARCHAR(255), sip_from_host VARCHAR(255), contact_user VARCHAR(255), contact_host VARCHAR(255), state VARCHAR(255), direction VARCHAR(255), user_agent VARCHAR(255), profile_name VARCHAR(255), hostname VARCHAR(255), contact VARCHAR(255), presence_id VARCHAR(255), presence_data VARCHAR(255), call_info VARCHAR(255), call_info_state VARCHAR(255), expires INTEGER default 0, status VARCHAR(255), rpid VARCHAR(255), sip_to_tag VARCHAR(255), sip_from_tag VARCHAR(255), rcd INTEGER not null default 0 ); -- -- table [sip_presence] -- CREATE TABLE sip_presence ( sip_user VARCHAR(255), sip_host VARCHAR(255), status VARCHAR(255), rpid VARCHAR(255), expires INTEGER, user_agent VARCHAR(255), profile_name VARCHAR(255), hostname VARCHAR(255), network_ip VARCHAR(255), network_port VARCHAR(6), open_closed VARCHAR(255) ); -- -- table [sip_authentication] -- CREATE TABLE sip_authentication ( nonce VARCHAR(255), expires INTEGER, profile_name VARCHAR(255), hostname VARCHAR(255), last_nc INTEGER ); -- -- table [sip_shared_appearance_subscriptions] -- CREATE TABLE sip_shared_appearance_subscriptions ( subscriber VARCHAR(255), call_id VARCHAR(255), aor VARCHAR(255), profile_name VARCHAR(255), hostname VARCHAR(255), contact_str VARCHAR(255), network_ip VARCHAR(255) ); -- -- table [sip_shared_appearance_dialogs] -- CREATE TABLE sip_shared_appearance_dialogs ( profile_name VARCHAR(255), hostname VARCHAR(255), contact_str VARCHAR(255), call_id VARCHAR(255), network_ip VARCHAR(255), expires INTEGER ); -- -- table [sip_recovery] -- CREATE TABLE sip_recovery ( runtime_uuid VARCHAR(255), profile_name VARCHAR(255), hostname VARCHAR(255), uuid VARCHAR(255), metadata text ); -- -- table [fifo_outbound] -- create table fifo_outbound ( uuid varchar(255), fifo_name varchar(255), originate_string varchar(255), simo_count integer, use_count integer, timeout integer, lag integer, next_avail integer not null default 0, expires integer not null default 0, static integer not null default 0, outbound_call_count integer not null default 0, outbound_fail_count integer not null default 0, hostname varchar(255), taking_calls integer not null default 1, status varchar(255), outbound_call_total_count integer not null default 0, outbound_fail_total_count integer not null default 0, active_time integer not null default 0, inactive_time integer not null default 0, manual_calls_out_count integer not null default 0, manual_calls_in_count integer not null default 0, manual_calls_out_total_count integer not null default 0, manual_calls_in_total_count integer not null default 0, ring_count integer not null default 0, start_time integer not null default 0, stop_time integer not null default 0 ); -- -- table [fifo_bridge] -- create table fifo_bridge ( fifo_name varchar(1024) not null, caller_uuid varchar(255) not null, caller_caller_id_name varchar(255), caller_caller_id_number varchar(255), consumer_uuid varchar(255) not null, consumer_outgoing_uuid varchar(255), bridge_start integer ); -- -- table [fifo_callers] -- create table fifo_callers ( fifo_name varchar(255) not null, uuid varchar(255) not null, caller_caller_id_name varchar(255), caller_caller_id_number varchar(255), timestamp integer ); -- -- table [voicemail_msgs] -- CREATE TABLE voicemail_msgs ( created_epoch INTEGER, read_epoch INTEGER, username VARCHAR(255), domain VARCHAR(255), uuid VARCHAR(255), cid_name VARCHAR(255), cid_number VARCHAR(255), in_folder VARCHAR(255), file_path VARCHAR(255), message_len INTEGER, flags VARCHAR(255), read_flags VARCHAR(255), forwarded_by VARCHAR(255) ); -- -- table [voicemail_prefs] -- CREATE TABLE voicemail_prefs ( username VARCHAR(255), domain VARCHAR(255), name_path VARCHAR(255), greeting_path VARCHAR(255), password VARCHAR(255) );

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conf call count
by Rodney 19 Dec '11

19 Dec '11

Mariusz, thankyou, that expression worked great! <action application="say" data="en number pronounced ${expr(${conf_count}-1)}"/> ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, December 18, 2011 7:17 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 88 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Weekly Conference Torrents. (Stephen Dame) 2. Re: Weekly Conference Torrents. (Stephen Dame) 3. Re: Where are table definitions of database for freewsitch (latest GIT version) (Jeff Lenk) 4. announce conf count (total callers minus 1)? (Rodney) 5. destination_number cleanup (Elliott Vogel) 6. Re: Grandstream GXP 2000 (Ted Schober) 7. Odp: announce conf count (total callers minus 1)? (Mariusz Czulada) Jeroen,Giovanni Thanks, I tried playing all three wav,ogg,mp3 at 3x in vlc still not understandable. This is on Windows 7, where itunes, media player do the same thing. I'll try playing direct from within freeswitch for now. Stephen -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Sunday, December 18, 2011 4:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weekly Conference Torrents. maybe the recordings are at 48khz and you're playing at 16khz? -giovanni On 12/18/11, Jeroen Eeuwes <jeroeneeuwes(a)gmail.com> wrote: > Hi Stephen, > >> Question. I have downloaded the last 4 weekly confrerecne calls from >> torrents, but all the files play in slow motion sound like lion roaring > > I never downloaded it before, but it sounds the same here. > > If you speed up the file to 300% it sounds OK. Under linux I used this > to play the file. For example: > > mplayer -speed 3 conf_call_2011-12-14.mp3 > > You can also use sox to get a file which sounds better. For example: > > sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 > > Beste regards, > Jeroen Eeuwes > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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 Just tried playing them direct in freeswitch use latest git... they sound the same not understandable by default I tried in normal, and cd quality 48,000 conf using conference 3300 play file.mp3 stephen -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jeroen Eeuwes Sent: Sunday, December 18, 2011 1:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weekly Conference Torrents. Hi Stephen, > Question. I have downloaded the last 4 weekly confrerecne calls from > torrents, but all the files play in slow motion sound like lion roaring I never downloaded it before, but it sounds the same here. If you speed up the file to 300% it sounds OK. Under linux I used this to play the file. For example: mplayer -speed 3 conf_call_2011-12-14.mp3 You can also use sox to get a file which sounds better. For example: sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 Beste regards, Jeroen Eeuwes _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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 All the required tables should be created on first run. Same as always. Is that not happening? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Where-are-table-definitions-o… Sent from the freeswitch-users mailing list archive at Nabble.com. this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? conf/dialplan/default/01_Announce_Conf_Count.xml: <include>  <extension name="Announce Conf Count (Inline)"> <condition field="destination_number" expression="^ANNOUNCE_CONF_COUNT_PRIVATE$"> <action application="set" data="conf_count=${conference ${conference_name} list count}"/> <action application="log" data="INFO ${conf_count}"/> <action application="say" data="en number pronounced ${conf_count}"/> </condition> </extension> </include> I was wondering if anyone has a regex expression that works to return just digest? I have some clients sending requests formatted to +1 (555) 555-5555, 555-555-5555, 555.555.5555 which aren’t be processed by our dial plan because we are expecting all numbers (5555555555) Jonas Gauffin <jonas.gauffin@...> writes: > > Hello > > I have problems with TCP and GXP2000 phones. > They work fine using UDP, but no calls arrive if I switch to TCP. > I've upgraded to the latest GS firmware today to see it that helped, > but it didn't. > > Have anyone else had problems with TCP? > > Regards, > Jonas > > If you touch the account page after setting it up and registering with FS, FS will then deny registration. I suspect the password gets messed up. If you need to change the account settings you need to factory reset and get it all right with one pass. What about <action application="say" data="en number pronounced ${expr(${conf_count}-1)}"/>? Regards, Mariusz Dnia 18-12-2011 o godz. 19:22 Rodney napisał(a): this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? > > > > >conf/dialplan/default/01_Announce_Conf_Count.xml: ><include> > ><extension name="Announce Conf Count (Inline)"> ><condition field="destination_number" expression="^ANNOUNCE_CONF_COUNT_PRIVATE$"> ><action application="set" data="conf_count=${conference ${conference_name} list count}"/> ><action application="log" data="INFO ${conf_count}"/> ><action application="say" data="en number pronounced ${conf_count}"/> ></condition> ></extension> ></include> _______________________________________________ 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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N-tier hierarchy real time billing using FS and NibbleBill
by Gael Martin 18 Dec '11

18 Dec '11

Hi, I need to develop a project with FreeSwitch that can bill calls in realtime not just for the end user but for all its parent account at different rate too and if one of them run out of credit the call stops. The NibbleBill module only allow to rate 2 account and 2 rates at the same time if we bill the A and B leg, this would be fine but I sometime need to bill 3 or more tier but realistically this is unlikely to go beyond 5 tiers. I tried a proof of concept by looping the call to FreeSwitch and NibbleBill each loop using it's own account and rate until there's no parent account for a user and then go out to the PSTN gateways, I cannot use the loopback system because it would remove all the middle calls once the 2 ends are connected together, so I use a sofia bridge to the local IP, (I just used a different sofia interface than the one the end user normally register into and this looping interface is not public) This is working fine and does exactly what I want, so far I've tested it with 5 tiers and as soon as one tier run out of credit it will disconnect the entire chain of call and adjust each account the correct amount. I have not tested with multiple end user calls at the same time but I expect it would work the same way. My only concern is performance and any potential issue doing some many loopback in terms of quality or timing. Looping calls in telephony system are normally reserved only as a last chance attempt to sort a problem. So I was wondering if someone had done something similar (i.e. looping calls through the same FreeSwitch instance) with no issues, and if there were anything to watch for or any tweaks to the sofia interface to make sure this will not cause an issue when scaling this system. I'm fairly new to FreeSwitch so I dont know by heart all the different options yet. Thanks. PS: I'm happy to share my proof of concept if anyone's interested doing the same.

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Problem with using mod_xml_curl at user authentication
by rakib 0000 18 Dec '11

18 Dec '11

Hello list, I'm a newbie FreeSwitch user, trying to learn how mod_xml_curl works. I've done the basics things mentioned in FreeSwitch wiki pages about using mod_xml_curl. Now my plan to use mod_xml_curl for user registration. So, what I've done is - I wrote a server end application which reads POST request from mod_xml_curl and tries to find whether it's a valid post request (REGISTER) or not. If the request is a REGISTER request - then the application extracts the user id, domain. Then, it queries the database and extracts password and creates xml response. One of my sample response are like below (the following response has been taken from /tmp): <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="192.168.0.108"> <groups> <group name="default"> <users> <user id="1100"> <params> <param name="password" value="sakib"/> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> With above response, I didn't had any luck. It fails to register, with following response: 2011-12-18 14:51:29.253474 [WARNING] sofia_reg.c:2446 Can't find user [1100(a)192.168.0.108] You must define a domain called '192.168.0.108' in your directory and add a user with the id="1100" attribute and you must configure your device to use the proper domain in it's authentication credentials. To figure it out, I've tried in different ways too. Like, I've used the above response patterns to create a user under conf/directory/default/ directory, but didn't work. Then I tried the following xml response pattern : <include> <user id="1003"> <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> </include> with the above xml I manage to create a user and authenticate perfectly. But, I've modified my server end application to give a that kind of response, but didn't work though. I've copied the server response (from the content dumped under /tmp) to create an user under conf/directory/default/ directory - but it works. So, my question is what am I missing here? FreeSwitch tries to get the response from the server, it gets the response too, but somehow not interpreting the response (or something else?) and gives "can't find user" error. Any hint's , suggestion will be very helpful. (note, I'm not using any webserver here.) Thanks in advance, Rakib

1 0

Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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Some questions about FS-2828
by ib-freeswitch＠bzsolutions.it 17 Dec '11

17 Dec '11

Hi, I'm studing FS-2828, why freeswitch statemachine always 'remap' 180 with SDP in a 183 ? I already ported FS-2828 patch to current GIT but my patch didn't forward new SDP when 180 is received after first 183 http://pastebin.com/dqiBcYC8 My idea is to create a patch to forward all 1XX messages to calling party. Thanks Igor-

1 0

The system cannot create any sessions at this time
by Stephen Wilde 17 Dec '11

17 Dec '11

Hi, what are the possible cause of the critical errors: 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session 2011-12-15 19:14:37.538948 [CRIT] switch_core_session.c:1775 The system cannot create any sessions at this time. 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system cannot create any sessions at this time. 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system cannot create any sessions at this time. 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session and so on. This kind of errors appear in a specific time interval (for example from 19:14:31 until 19:14:39) and after this time interval all continue to work normally. This is not a "max_sessions" related problem because in this case appear a "Over Session Limit!" message. I think also that this is not a "min idle" problem because I have not set this parameter, my status is: UP 0 years, 0 days, 15 hours, 26 minutes, 32 seconds, 182 milliseconds, 131 microseconds FreeSWITCH is ready 4279392 session(s) since startup 3463 session(s) 30/100 6000 session(s) max min idle cpu 0.00/77.00 What are the possible causes of this message?

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Transfer a call to other node
by Mariusz Czulada 17 Dec '11

17 Dec '11

Hi all, Looking at Tech-Invite site I found such flow example: http://www.tech-invite.com/Ti-sip-service-06.html I wonder if it's (currently) possible to implement a replacing INVITE using FreeSWITCH (for Bob and Carol sides). Regards, Mariusz

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