March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

user_data sends two header_strings named key
by Leon de Rooij 25 Mar '09

25 Mar '09

Hi, I think this is a bug in mod_commands.c : When using "user_data" function from mod_commands (mod_commands.c, line 358) A header string "key" is added to params (mod_commands.c, line 362) Then switch_xml_locate_user is called (switch_xml.c, line 1712) And that function adds another header string "key" (switch_xml.c, line 1740) This results in two keys with the same name being sent ( to in my case mod_xml_curl ). I opened an issue in jira for this (is it alright like that ?) : http://jira.freeswitch.org/browse/MODAPP-242 regards, Leon

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XML (curl) returned for Event-Calling-Function = switch_xml_locate_user
by Leon de Rooij 25 Mar '09

25 Mar '09

Hi, I'm trying to get some cli commands working in combination with xml- curl. Endpoints are parsed properly for SIP registrations and invites, but when I use the CLI command "user_exists" it returns false, while I do return an endpoint (same syntax as for a sofia_reg_parse_auth event) on the webserver. Should switch_xml_locate_user event receive a different syntax ? freeswitch@internal> user_exists accountcode gigaset toyos.nl false XML returned from webserver: <?xml version="1.0" encoding="UTF-8"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="toyos.nl"> <user id="gigaset" mailbox=""> <params> <param name="a1-hash" value="a211336e29c4f3756e6af343ce6da27b"/> <param name="email-addr" value="test(a)test.com"/> <param name="vm-delete-file" value="true"/> <param name="vm-attach-file" value="true"/> </params> <variables> <variable name="accountcode" value="gigaset"/> <variable name="caller_id_name" value="gigaset"/> <variable name="caller_id_number" value="1000"/> <variable name="email_addr" value="test(a)test.com"/> <variable name="privacy" value="no"/> <variable name="process_cdr" value="true"/> <variable name="sip_secure_media" value="false"/> <variable name="country_code" value="31"/> <variable name="area_code" value="320"/> </variables> </user> </domain> </section> </document> thanks & regards, Leon

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FreeSWITCH Chinese Community
by dujinfang 25 Mar '09

25 Mar '09

Hi all, In about a year of playing with FreeSWITCH, I really like it. Not only the software is great but also the community. As people keep asking me where can find some documentation in Chinese, I told them if they want to play in deep they need to know English better. However, the fact is, even one can read English without problem, he still prefer find some information in his native language first. So, I'd like to start a Chinese community hoping FreeSWITCH can get more popular in China. As we saw big telecom companies merged in the last year, I bet VoIP technology and application will go more and more faster in the near future... While FS official site has plenty of documentation, obviously we don't want to translate word by word. Then where should we start from? A forum? There was a big argue between a forum and mailing list in the last few weeks, and finally an English forum and an Italian one is out. While I can find server in China, the major pain is that running any kind of BBS in China mainland need to get some kind of permissions by the government first. Any idea, suggestion? Anyone want to help or cooperate about this? And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn ? Best regards, Seven.

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ANNOUNCEMENT: ClueCon 2009 - Register Today!
by Michael Collins 25 Mar '09

25 Mar '09

The FreeSWITCH Team is pleased to announce that all may register for ClueCon 2009 immediately! ClueCon is the Telephony Developers Conference by developers, for developers. This year's event will be held at the beautiful Wyndham Hotel in Chicago, August 4-6. Special room rates have been secured and you will enjoy staying at this luxurious hotel. The conference attendee price is only $499 per person, but hurry! This early bird special price is valid only until June 30th. The schedule is still being finalized, however you can expect to hear from notable names in the world of telephony. You will also have opportunities to spend one-on-one time with developers, vendors, and other attendees. One special highlight for this year is the great MacBook Pro giveaway! On Thursday at the end of the conference a 15" MacBook Pro will be given away to one lucky attendee. This laptop will be laser engraved with the logos of all of the sponsors for ClueCon 2009 and will be used as the presenters' laptop for the entire conference. Start making your travel plans right away. Visit http://www.cluecon.com and sign up today. We look forward to seeing you in Chicago this August! Michael S Collins 877-742-CLUE

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sip cancel request fails
by Steven Ward 24 Mar '09

24 Mar '09

A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work. Why is this request resulting in a 481? I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console: recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: ------------------------------------------------------------------------ CANCEL sip:70904@b-pbx-lab-1.mynet.net<sip%3A70904(a)b-pbx-lab-1.mynet.net>SIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport From: "Steve" <sip:70904@10.1.21.44 <sip%3A70904(a)10.1.21.44> >;tag=as7f6965ea To: <sip:70904@b-lab-1.mynet.net <sip%3A70904(a)b-lab-1.mynet.net>> Call-ID: 237598fd102b739a03b4a4047bf69843(a)10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 From: "Steve" <sip:70904@10.1.21.44 <sip%3A70904(a)10.1.21.44> >;tag=as7f6965ea To: <sip:70904@b-lab-1.mynet.net <sip%3A70904(a)b-lab-1.mynet.net> >;tag=71m745HKHKyjc Call-ID: 237598fd102b739a03b4a4047bf69843(a)10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------- Thanks. - SW

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Sip for Skype
by Dan 24 Mar '09

24 Mar '09

You probably already saw this but.... http://www.skypeforsip.com/ Skype is supporting sip for business users.

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sip cancel request fails
by Steven Ward 24 Mar '09

24 Mar '09

A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work. Why is this request resulting in a 481? I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console: recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: ------------------------------------------------------------------------ CANCEL sip:70904@b-pbx-lab-1.mynet.net<sip%3A70904(a)b-pbx-lab-1.mynet.net>SIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport From: "Steve" <sip:70904@10.1.21.44 <sip%3A70904(a)10.1.21.44> >;tag=as7f6965ea To: <sip:70904@b-lab-1.mynet.net <sip%3A70904(a)b-lab-1.mynet.net>> Call-ID: 237598fd102b739a03b4a4047bf69843(a)10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 From: "Steve" <sip:70904@10.1.21.44 <sip%3A70904(a)10.1.21.44> >;tag=as7f6965ea To: <sip:70904@b-lab-1.mynet.net <sip%3A70904(a)b-lab-1.mynet.net> >;tag=71m745HKHKyjc Call-ID: 237598fd102b739a03b4a4047bf69843(a)10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------- Thanks. - SW

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[SUSE Enterprise/FS 1.0.3] ODBC-related compile error
by Gilles 24 Mar '09

24 Mar '09

Brian West > We changed this to autodetect it means you have the libs but no devel headers.. either disable it or install the dev headers. Thanks guys, that was it: The stock SLED 10 that I have on this MSI host did have unixODBC.rpm but not its unixODBC-devel.rpm counterpart. I re-read the "Download & Installation Guide" page, which doesn't say that Freeswitch will compile for ODBC instead of SQLite if it detects unixODBC... but doesn't check that its counterpart unixODBC-devel is also installed :-/ I suggest that either the config/make script be updated to check for this, or the wiki be edited to tell users to check for both packages (if N.A. : yast install unixODBC-devel OR rpm -UVH unixODBC-devel.rpm) or configure Freeswitch to use SQLite instead of ODBC (good enough to get newbies started). Cheers,

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[SUSE Enterprise/FS 1.0.3] ODBC-related compile error
by Gilles 23 Mar '09

23 Mar '09

Hello, This is my first try at compiling Freeswitch (1.0.3). Since I have a Linux box lying around, I'm giving it a shot on a "SUSE Linux Enterprise Desktop 10 SP1 (i586)". Per the instructions in the wiki, I run "make", but get the following error : http://pastebin.ca/1369381 According to the archives, it's likely due to the missing ODBC-devel package. I have a couple of questions: 1. Does Freeswitch require ODBC? 2. Please forgive the newbie question, but... how do I find and install this package on Suse? Thank you.

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Inbound calls
by Леша... 23 Mar '09

23 Mar '09

Hi all! There is an error if I try to make inbound call. I don't think it's hard but I'm tired solving it. Any time I call my city number, FS always returns me an error [INCOMPATIBLE_DESTINATION] 2009-03-23 00:29:40 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/mydomain/number-ani@ip [8d0dcff8-1728-11de-b150-e189b327b8ae] 2009-03-23 00:29:40 [NOTICE] sofia.c:2927 sofia_handle_sip_i_state() Hangup sofia/mydomain/number-ani@ip [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-03-23 00:29:40 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 2 (sofia/mydomain/number-ani@ip) Ended 2009-03-23 00:29:40 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/mydomain/number-ani@ip [CS_HANGUP] My extension is: <extension name="Inbound-number"> <condition field="destination_number" expression="^dest_number$"> <action application="ring_ready" />  <action application="bridge" data="user/1000@$mydomain"/> </condition> </extension> I've made it all the same as http://wiki.freeswitch.org/wiki/Home_PBX_Example (with my own diference line inbound number, dest_number , etc) Could you share some expirience? Give any solution? Or link where I could learn more about it? =\

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