June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Are you using NDLB-force-rport = "safe" ?
by Michael Collins 25 Jun '12

25 Jun '12

Hi gang, A friend of mine asked how many people are using this. Are you? It's mostly for Polycoms and non-polys to have force rport in a typical hosted environment. If you have been using this setting in production please give us some feedback. Thanks! -Michael

1 0

is it possible to use lua in mod_xml_curl ?
by Samira Mh 24 Jun '12

24 Jun '12

the supported languages in mod_xml_cur include: perl,python,php,c++,c# , but can i use lua ? lua is meet our requirements for implementing mod_xml_curl

3 3

How to break a playback with loops
by Daniel Liang 24 Jun '12

24 Jun '12

Hi, I am trying to playing back a wav file a few times, and on receiving a dtmf event, it stops playing the file. The command I sent is: sendmsg call-command: execute execute-app-name: playback execute-app-arg: <the wav file> loops: 4 On receiving a dtmf event, I send a break command: sendmsg call-command: execute execute-app-name: break However, that only breaks one loop of the playback. And I have to press a dtmf for 4 times to stop the file. I tried sending 4 break commands all at once. But it didn't work. And even worse, my program stops receiving dtmfs anymore. My guess is that the first break stops the first loop, but before the second loop starts, the following 3 breaks stop something else that I don't know what they are. Any help is appreciated. Thanks. Daniel Liang

6 6

Transport parameters when using RFC3327 Path and fs_path
by Richard Brady 24 Jun '12

24 Jun '12

Hi folks I have FreeSWITCH running behind Kamailio, which acts as a TCP (client side) to UDP (FreeSWITCH side) bridge. The client registers with a binding like: Contact: <sip:1010@192.168.1.228:32204;transport=tcp> and the Kamailio server is adding a Path header as follows: Path: <sip:10.1.1.1;lr> FreeSWITCH stores this as: Contact:<sip:1010@192.168.1.228:32204;transport=tcp;fs_path=%3Csip%3A10.1.1.1%3Blr%3E> But then when originating /bridging a call to this user, I get: [DEBUG] sofia_glue.c:2055 sip:10.1.1.1;lr;transport=tcp Setting proxy route to sofia/internal/sip:1010@192.168.1.228:32204 which indicates that the wrong transport is being used. If I add transport=udp to the Path header generated by Kamailio, the correct transport is used. However the Path URI, by not including a transport parameter, implicitly specifies UDP. Am I doing something wrong or could this possibly be a bug? Regards, Richard

1 0

Changing FROM header
by Shishko 24 Jun '12

24 Jun '12

Yes Paul, I tried, and it works now. Thanks!

1 0

Sending NOTIFY with perl ESL
by Vallimamod ABDULLAH 24 Jun '12

24 Jun '12

Hello List, I am trying to change BLF lamps status by sending PRESENCE_IN event via perl ESL. My event is sent correctly but corresponding notify always have an empty <dialog-info> wether Answer-State is confirmed or early. My script is like this: my $e = ESL::ESLevent->new("PRESENCE_IN"); $e->addHeader("proto", "sip"); $e->addHeader("from", "51(a)10.0.0.1"); $e->addHeader("login", "51(a)10.0.0.1"); $e->addHeader("event_type", "presence"); $e->addHeader("alt_event_type", "dialog"); $e->addHeader("event_count", "1"); $e->addHeader("Presence-Call-Direction", "outbound"); $e->addHeader("Answer-State", "early"); $con->sendEvent($e); The corresponding ESL event as received by freeswitch is: sendevent PRESENCE_IN Event-Name: PRESENCE_IN proto: sip from: 51(a)10.0.0.1 login: 51(a)10.0.0.1 event_type: presence alt_event_type: dialog event_count: 1 Presence-Call-Direction: outbound Answer-State: early But the SIP NOTIFY sent to the subscribers is: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="563" state="partial" entity="sip:51@10.0.0.1"> </dialog-info> i.e. it does not contain any <dialog> tag. Is there something wrong with my script ? Thank you! -- Best Regards, - vma .

1 0

question on mod_lua for serving_configurations(implement realtime in freeswitch)?
by Samira Mh 24 Jun '12

24 Jun '12

hi, in mod_lua for Serving_Configuration ( http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration ) i have defined the settings as follows: 1- <configuration name="lua.conf" description="LUA Configuration"> <settings> <param name="xml-handler-script" value="dp.lua"/> <param name="xml-handler-bindings" value="directory"/> </settings> </configuration> 2- /usr/local/freeswitch/scripts/dp.lua freeswitch.consoleLog("INFO", "SECTION " .. XML_REQUEST["section"] .. "\n") freeswitch.consoleLog("INFO", "TAG_NAME " .. XML_REQUEST["tag_name"] .. "\n") freeswitch.consoleLog("INFO", "KEY_NAME " .. XML_REQUEST["key_name"] .. "\n") freeswitch.consoleLog("INFO", "KEY_VALUE " .. XML_REQUEST["key_value"] .. "\n") freeswitch.consoleLog("INFO",params:serialize()) io.write("TEST \n" .. params:serialize("xml") .. "\n"); --io.write("SIP_AUTH_METHOD\n" .. params:getHeader("sip_auth_method") .. "\n"); --io.write("sip_profile\n" .. params:getHeader("sip_profile") .. "\n"); local sipprofile = params:getHeader("sip_profile"); print ("sip profile is:" .. sipprofile); 3- the output of the above lua file is : TEST <event> <headers> <Event-Name>REQUEST_PARAMS</Event-Name> <Core-UUID>1d640ebe-be01-11e1-b2cd-4109894802cc</Core-UUID> <FreeSWITCH-Hostname>PBX</FreeSWITCH-Hostname> <FreeSWITCH-Switchname>PBX</FreeSWITCH-Switchname> <FreeSWITCH-IPv4>192.168.10.89</FreeSWITCH-IPv4> <FreeSWITCH-IPv6>%3A%3A1</FreeSWITCH-IPv6> <Event-Date-Local>2012-06-24%2018%3A13%3A48</Event-Date-Local> <Event-Date-GMT>Sun,%2024%20Jun%202012%2013%3A43%3A48%20GMT</Event-Date-GMT> <Event-Date-Timestamp>1340545428075826</Event-Date-Timestamp> <Event-Calling-File>sofia_reg.c</Event-Calling-File> <Event-Calling-Function>sofia_reg_parse_auth</Event-Calling-Function> <Event-Calling-Line-Number>2364</Event-Calling-Line-Number> <Event-Sequence>1240</Event-Sequence> <action>sip_auth</action> <sip_profile>internal</sip_profile> <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent> <sip_auth_username>1000</sip_auth_username> <sip_auth_realm>192.168.10.89</sip_auth_realm> <sip_auth_nonce>5fef3276-be02-11e1-b2fd-4109894802cc</sip_auth_nonce> <sip_auth_uri>sip%3A192.168.10.89</sip_auth_uri> <sip_contact_user>1000</sip_contact_user> <sip_contact_host>192.168.18.120</sip_contact_host> <sip_to_user>1000</sip_to_user> <sip_to_host>192.168.10.89</sip_to_host> <sip_from_user>1000</sip_from_user> <sip_from_host>192.168.10.89</sip_from_host> <sip_request_host>192.168.10.89</sip_request_host> <sip_auth_qop>auth</sip_auth_qop> <sip_auth_cnonce>8725f832cb29a466ba073ef5d63de959</sip_auth_cnonce> <sip_auth_nc>00000010</sip_auth_nc> <sip_auth_response>ee0155cd5b41536e56898ff585502e75</sip_auth_response> <sip_auth_method>REGISTER</sip_auth_method> <key>id</key> <user>1000</user> <domain>192.168.10.89</domain> <ip>192.168.18.120</ip> </headers> </event> sip profile is :internal --------------------------------------------------- Now i am going to check using reltime every user that registers on sip phone -like 3CXphone - instead of in-memory method (flat file on freeswitch), now what is the next step for implementing? Now need to define mysql and insert query in lua script and compare the query with the value of params:getHeader ? the resource for mod_lua serving configuration is a bit plz help, thanks so much

2 1

internal extension is unregistered when registering from remote location
by Juan Pablo L. 24 Jun '12

24 Jun '12

Hi, i have an internal extension 2000 registering from a remote location, extension 2000 can make calls to internal extension and through gateways but nobody can call the 2000 extension, when someone tries to reach extension 2000 there is this error: "switch_ivr_originate.c:2492 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]" and voicemail invites to leave a message. when extension 2000 registers i get this in the FS logs: sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal-hq' for [2000(a)10.19.0.8] from ip 190.242.32.118 so it is registering, when i check in the database this extension is not there. i would appreciate any hints. thanks!

2 3

Changing FROM header
by Shishko 24 Jun '12

24 Jun '12

Hi, is there a way I can change From header in bridge application line? In a-leg I get something as "xxxxxx <sip:xxxxxx@domain.tld>", and I'd like to somehow change it to "yyyyyy <sip:yyyyyy@domain.tld>" in b-leg. I tried with sip_h_From and sip_from_uri, but no success... Thanks

2 1

persistent registration
by Darcy Primrose 24 Jun '12

24 Jun '12

I have mutliple_registrations set to 'true", if a user on a dsl gets rebooted and receives a new IP address, the old registration remains persistent and gets a value of EXPSECS(-xxxx) and this continues unless it is flushed or if the command 'RELOADXML" is executed. Is there anyway to clear these negative (persistent) registrations with intervention. Darcy Primrose

1 0