March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

doublenat outgoing call issue
by Bruce Hopkins 15 Mar '10

15 Mar '10

Hi, I wonder if some kind soul might be able please to help me by pointing out what I am doing wrong with my doublenat configuration. I have incoming calls from WAN --> LAN working fine, but when I try to call from LAN -->WAN, the Freeswitch CLI says: "[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]" The setup is essentially: phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z) Phone_1 is able to register with Freeswitch using a "doublenat" sip profile I set up using port 5090, created from the default external profile with the port changed. phone_1 can initiate calls to phone_2 fine with no problems with audio - everything works just fine in that direction. However when I try to call from phone_2 --> phone_1 I get the message "The person at extension 1002 is not available. Record your message at the tone." Meanwhile the CLI shows that Freeswitch says phone_1 is not registered. In trying to solve this I have run a Wireshark trace onthe PC running the phone_1 softphone to check that the SIP registration seems to go as I understand it should. the dialog is as follows: phone_1 --> FS request: REGISTER FS --> phone_1 status: 401 Unauthorised phone_1 --> FS request: REGISTER FS --> phone_1 status: 200 OK phone_1 --> FS request: SUBSCRIBE FS --> phone_1 status: 405 Method Not Allowed There does appear to be a difference here to what happens when phone_2 (on the same LAN as Freeswitch) registers. With phone_2 I see: phone_2 --> FS request: REGISTER FS --> phone_2 status: 401 Unauthorised phone_2 --> FS request: REGISTER FS --> phone_2 status 200 OK phone_2 --> FS request: SUBSCRIBE FS --> phone_2 status: 202 Accepted FS --> phone_2 request: NOTIFY phone_2 --> FS status: 200 OK More configuration details: In conf/sip_profiles/doublenat.xml I have not uncommented <param name="force-register-domain" value="$$(domain)"/> In conf/sip_profiles/doublenat.xml I have tried setting the context to each of public and default. This does not seem to make a difference. In conf/directory/doublenat.xml I have set the domain name to be the domain name registerd for Fs's external IP with DNS, and used by phone_1. In conf/directory/doublenat.xml I have tried both of the following 9I don't know if this is relevant) 1. Firstly with phone_1 registering to an extension in the default configuration, held in conf/directory/default/: <groups> <group name="default"> <users> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </users> </group> 2. Then I tried moving the .xml file for phone_1's extension to conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show: <groups> <group name="doublenat"> <users> <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> </users> </group> Unfortunately this does not make a difference to the issue though. If anybody had the time to tell me what I am doing wrong I would be hugely grateful ! Many thanks in advance Bruce

4 18

regarding the recent sangoma boost
by Tzury Bar Yochay 15 Mar '10

15 Mar '10

one must run # mkdir /root/sangoma before head. otherwise, sangoma_prid will fail to start I discoverd this since /var/log/sangoma_mgd.log file was empty even though the line local2.* /var/log/sangoma_mgd.log exists in /etc/syslog.conf I did the following: # export LD_LIBRARY_PATH=/usr/lib/sangoma_prid # /usr/sbin/sangoma_prid &> sangoma_prid.log an then with # tail -f sangoma_prid.log I got ==> sangoma_prid.log <== ================System restart============= = Sangoma PRI Protocol Stack Daemon = = Version: 1.58 = = Date: Feb 18 2010 = = Wanpipe Release: wanpipe-3.5.10.3 = = Revision:Revision: 15364 = =========================================== sh: gen_fs_gcore.sh: not found touch: cannot touch `/root/sangoma/sangoma_prid.gcore.24821': No such file or directory /usr/bin/gcore: 83: cannot create /root/sangoma/sangoma_prid.gcore.24821: Directory nonexistent gcore: failed to create /root/sangoma/sangoma_prid.gcore.24794 I then ran `# mkdir /root/sangoma` and restart everything Right now I am facing another problem which hope to over come right after lunch the problem is this error freeswitch@snaptrunk-02> originate openzap/1/a/0525133399 XML public 2010-03-15 12:20:56.307066 [WARNING] ozmod_sangoma_boost.c:341 TX EVENT: CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[N/A] Cd=[0525133399] Ci=[N/A] Rdnis=[] 2010-03-15 12:21:06.858887 [WARNING] sangoma_boost_client.c:221 TX EVENT (N): CALL_START_NACK:(82) [w1g1] Rc=0 CSid=1 Seq=1 2010-03-15 12:21:06.858887 [ERR] switch_ivr_originate.c:2422 Cannot create outgoing channel of type [openzap] cause: [RECOVERY_ON_TIMER_EXPIRE] -ERR RECOVERY_ON_TIMER_EXPIRE 2010-03-15 12:21:06.858887 [DEBUG] switch_ivr_originate.c:3220 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] freeswitch@snaptrunk-02>

1 0

Skypiax has been renamed as Skypopen
by Giovanni Maruzzelli 15 Mar '10

15 Mar '10

Ciao FreeSWITCHers, is with pleasure that we're announcing the renaming of mod_skypiax module to mod_skypopen. The new wiki page is at http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk To upgrade from a previous Skypiax installation, please read: http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Upgra… Many many many thanks to the wonderful FreeSWITCH team that make it all happens! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

2 2

LUA and LUASQL Install
by Aloysius Lloyd 15 Mar '10

15 Mar '10

Hi All, Can some one help me to install Lua . OS: CentOS 5.4 wget http://www.lua.org/ftp/lua-5.1.4.tar.gz tar zxf lua-5.1.4.tar.gz cd lua-5.1.4 make linux But getting error when running *make linux * * * *Here is the Output* In file included from lua.h:16, from lua.c:15: luaconf.h:275:31: error: readline/readline.h: No such file or directory luaconf.h:276:30: error: readline/history.h: No such file or directory lua.c: In function âpushlineâ: lua.c:182: warning: implicit declaration of function âreadlineâ lua.c:182: warning: assignment makes pointer from integer without a cast lua.c: In function âloadlineâ: lua.c:210: warning: implicit declaration of function âadd_historyâ make[2]: *** [lua.o] Error 1 make[2]: Leaving directory `/usr/src/lua-5.1.4/src' make[1]: *** [linux] Error 2 make[1]: Leaving directory `/usr/src/lua-5.1.4/src' make: *** [linux] Error 2 Thank you in advance. Thanks, Lloyd

3 6

mod_nibblebill and dynamic conference
by Alfredo Quiroga-Villamil 14 Mar '10

14 Mar '10

All: We are currently exploring the possibility of using mod_nibblebill and integrating it with a conference. It seems like it offers all the features and more that we need. The way we create a conference now is as follows: originate sofia/gateway/url/XXXXXXXXXXX &conference(myConf+flags{moderator,wait-mod}) >From the mod_nibblebill wiki page it seems like we need to set a few variables for the B leg: enable_heartbeat_events nibble_rate nibble_account= I was wondering if there is a way to set these variables via the API call shown above where the conference is created. If not, what options do you recommend to accomplish this? Thanks in advance, Alfredo

3 7

Trouble with P-Asserted-Identity header(s)
by Roland Hänel 14 Mar '10

14 Mar '10

Hello, I'm developing a very simple call-through application using FreeSWITCH, but currently I'm stuck because my provider requires me to send a 'P-Asserted-Identity' header for outgoing calls if I want so signal an arbitrary caller ID in the 'From' field. Setup is as follows: - a-leg comes in from an external gateway - dialplan dispatches the call to a perl handler script - perl script looks like this (simplified as much as possible for this description): $num = "02222222"; $session->answer(); $session->setVariable("effective_caller_id_name", "The Redirector"); $session->setVariable("sip_h_P-Asserted-Identity", '< sip:0123456789@q-loud.net <sip%3A0123456789(a)q-loud.net>>'); $session->execute("bridge", "sofia/gateway/mygateway/$num); The problem is now that FreeSWITCH correctly insertes a P-Asserted-Identity header as set in the perl script. But there is already an existing P-Asserted-Identity header by default, so I end up with two of them. This is the outgoing INVITE (b-leg): INVITE sip:02222222@x.x.x.x SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N Max-Forwards: 63 From: "The Redirector" <sip:01111111@q-loud.net <sip%3A01111111(a)q-loud.net> >;tag=DBSvZ9NZc750g To: <sip:02222222@x.x.x.x> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02 CSeq: 128071597 INVITE Contact: <sip:gw+mygateway@x.x.x.x:5080;transport=udp;gw=mygateway> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 244 X-Port: 5060 X-FS-Support: update_display P-Asserted-Identity: "The Redirector" <sip:01111111@q-loud.net<sip%3A01111111(a)q-loud.net> > P-Asserted-Identity: <sip:0123456789@q-loud.net<sip%3A0123456789(a)q-loud.net> > [... sdp ...] Numbers are: 01111111 - the original caller (a-leg & b-leg From:) 02222222 - the number redirected to (b-leg To:) 0123456789 - the number my provider needs in P-Asserted-Identity ('redirector's number') So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789 line, but unfortunately it already includes another P-Asserted-Identity 011111111 line, and that breaks the setup at my provider. I really appreciate any help. Greetings, Roland -- Roland Haenel QSC AG - http://www.qsc.de

4 6

Call transfer with Polycom works ok but a call is left on hold?
by Yehavi Bourvine 14 Mar '10

14 Mar '10

Hello, We have a strange behavior which my users told me today, but I have a feeling it lasts for a long time (we are running now version 16972). User A calls user B on a Polycom. User B press the "transfer" key and calls C who answers. B then press "transfer" again, and now A talks to C. However, B is left with a held call to the voicemail of C... Anyone saw this behaviour? unattended transfer and conference work ok. Thanks! __Yehavi:

2 3

how to access session variables after hangup???
by rentmycoder rentmycoder 14 Mar '10

14 Mar '10

Hi guys, Freeswitch rocks! except the documentation:( I'm not able to figure out how to access session variables after hangup... I try to dial out on a gateway and detect it the called party hangs up before bridge... bgapi originate {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun hangup.lua ${uuid}'}sofia/gateway/phoneno &park() I cannot access the variables using: 1. script arguments: 'luarun hangup.lua ${uuid}'} 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data (${api_hangup_hook} contains a variable) 2. session:getVariable("originate_disposition") 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt to call global 'getVar' (a nil value) 3. params:getHeader("variable_sip_req_uri") 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt to index global 'params' (a nil value) I have asked this before on the dev list too, but Anthony didn't gave me a clean answer here: http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html "In the script you can access the data from the env event which is automatically created for you and it contains all the channel variables from that departed channel." Ok, but how????? Please help...

2 1

js.dll crash on Windows Server 2008 R2 x64
by Quentin CALVEZ 14 Mar '10

14 Mar '10

Hello everybody, I've been trying Freeswitch those last days and there is one error that I don't know what to do about. I'm using FusionPBX to manage my freeswitch installation and it works great, however, whenever I try to use a function involving javascript (auto atendants, hunt groups...) the server crashs telling me that the faulty module is js32.dll. I really don't know what to do about it. Could someone help me :) ? Regards. Quentin Calvez

2 1

Inline and cond
by Dome Charoenyost 14 Mar '10

14 Mar '10

I try to use cond in my dialplan <condition field="destination_number" expression="^(1111)"> <action application="answer"/> <action inline="true" application="set" data="b=25"/> </condition> <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$"> <action application="log" data="ALERT ${b} is indeed greater than 12"/> <anti-action application="log" data="ALERT ${b} is not greater than 12"/> </condition> it's work but when use set b=${b1} it's not work <condition field="destination_number" expression="^(1111)"> <action application="answer"/> <action inline="true" application="set" data="b1=25"/> <action inline="true" application="set" data="b=${b1}"/> </condition> <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$"> <action application="log" data="ALERT ${b} is indeed greater than 12"/> <anti-action application="log" data="ALERT ${b} is not greater than 12"/> </condition> What's wrong ? inline or cond ? Dome C.

1 0