September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_perl won't load
by Nic Wolff 03 Sep '08

03 Sep '08

I'm new to Freeswitch and trying to start a clean install (no flags or options) with a stock modules.conf except mod_perl is enabled (on an EC2 Fedora instance if that matters!) and getting this error on startup: 2008-09-02 13:02:27 [CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error Loading module /usr/local/ freeswitch/mod/mod_perl.so **/usr/local/freeswitch/mod/mod_perl.so: undefined symbol: Perl_Tstack_base_ptr** Has anyone else seen this? Is anyone else running Freeswitch with mod_perl in EC2? Or, generally, what should I do next? Thanks - Nic

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NOOB QUESTION - Remote Extension Routing
by Shawn Lewis 03 Sep '08

03 Sep '08

I have a number of remote extensions connecting to my server. After some research I was able to get them register by creating a "doublenat.xml" in the sip-profile. So now I have users being able to register. Though in the examples I saw, the sample for being able to extension dial from an internal registered phone to them is not so easy, and that is where my question is. So I have extension 1001 registered locally. I have extension 2010 registered remotely. I have extension 2011 registered remotely. 2010 and 2011 both come from different locations. In my default.xml dialplan, how would I setup being able to dial 2010 or 2011 without KNOWING the IP address of their gateway? I have tried sofia/doublenat/2010 - doesnt work I have tried sofia/doublenat/$1%$$x.x.x.x - where x.x.x.x = my extern IP - doesnt work I have tried a manual sofia/doublenat/2010(a)x.x.x.x = where x.x.x.x = their NAT GW, and that does work But I will not always know their GW address........ Thanks Shawn ----- -- The information contained in this e-mail and any attached documents may be privileged, confidential and protected from disclosure. If you are not the intended recipient you may not read, copy, distribute or use this information. If you have received this communication in error, please notify the sender immediately by replying to this message and then delete it from your system.

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Originating Calls, without knowing the endpoint URL?
by Brian Wood 02 Sep '08

02 Sep '08

I'm experimenting with FreeSWITCH. I have a setup working with FXO, FXS, and SIP (to hardware and software phones, and gateways). I'd like to be able to place a new outgoing call and then do something, for example, play a message notifying me that the power has gone out. I have been successful doing this through the event socket, XML-RPC, JavaScript, etc. However, I have to know the full endpoint URL. I've searched the list, wiki, web for examples. I've seen several attempts at an example, but no clear and complete example. What I'd really like to be able to do is something like (JavaScript example): var s = new Session(); s.originate(session, "1234567890"); s.streamFile("test.wav"); Instead of: var s = new Session(); s.originate(session, "sofia/internal/1000"); s.streamFile("test.wav"); I would like to be able to originate a new session, and have FreeSWITCH lookup the endpoint. Transfer and bridge don't seem quite right, and exceute_extension seems to block (and require a existing session). I've looked at Session.Originate, but I can't seem to get it to do anything useful. So far, about the only ideas I have are to either parse the dialplan myself, or use mod_xml_curl to fetch the dialplan from my own software/database/etc. Then I could query my external source for the endpoint URL. This seems like overkill, but I can't find anyway to lookup the endpoint URL or place a call without one. I plan on adding more gateways and possibly T1 cards into the system, so I would prefer if FreeSWITCH could do the "routing", and my scripts not have to know what the endpoint is. It seems like there should be a way to do this (or at least a way query the for the endpoint given an extension/number). Maybe I'm just looking at the problem the wrong way. Maybe the dialplan is the complete wrong place to be looking, since it can do a lot more than just specify an endpoint. Is there any easy way to do this, or do I simply have to have a endpoint URL? On a related note, on an OpenZAP channel, how can I tell FreeSWITCH to use any available FXO channel? I can dial out using openzap/1/1/1234567890. However, if I have a T1 board or X number of FXO ports, is there a way to tell it to automatically use any available channel in a endpoint URL? Thanks.

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ANNOUNCEMENT: Documentation conf call Wednesday 9/3/2008 1pm Central
by Michael Collins 02 Sep '08

02 Sep '08

FYI, Here's your link: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_09_03 Thanks to lanman for getting this ready. FYI, I will not be on the call tomorrow because I will be at the orthodontist... I will check in on Thursday to see how it all went. -MC

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Re: [Freeswitch-users] DTMF problem
by Luis Manuel Zuccolo 02 Sep '08

02 Sep '08

Hi, Yes, I'm sure that the calls is marking the freeSWITCH extension. In asterisk I can see the digits sended. I'll go to increase the sleep time and see the results. Anyway, how can I use the app read? Would somebody show an example about this? Thanks in advance. Luis Zuccolo ----- Mensaje original ---- De: Michael S Collins <msc(a)freeswitch.org> Para: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Enviado: martes 2 de septiembre de 2008, 2:09:18 Asunto: Re: [Freeswitch-users] DTMF problem Are you sure that the call is making it to your extension? Also, you sleep for only 250 ms after start_dtmf and then issue a hangup. That's not a whole lot of time to send a digit. Perhaps your test would work better with the dialplan app "read" but that's just a guess on my part. -MC Sent from my iPhone On Sep 1, 2008, at 8:01 PM, "Luis M. Zuccolo" <luismzuccolo(a)yahoo.com.ar> wrote: > Hello, > > I've connected an analog pbx to FreeSWITCH throught a fxo port. > In this setup only need the voicemail functionality of freeSWITCH who > recived a dtmf from a pbx. > To test dtmf app i've created the > extension .../conf/dialplan/extensions/2.xml: > > <include> > <extension name="incoming-fx0-channel-2"> > <condition field="source" expression="mod_openzap"/> > <condition field="destination_number" expression=="^(2)$"> > <action application="answer"/> > <action application="start_dtmf" data="true"/> > <action application="sleep" data="250"/> > <action application="hangup"/> > </condition> > </extension> > </include> > > At console I don't see any reference to dtmf. > > Does anybody have any recommendations about this? > > Thanks in advance > Luis Zuccolo > > __________________________________________________ > Correo Yahoo! > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar > > > _______________________________________________ > 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 _______________________________________________ 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 Yahoo! Cocina Recetas prácticas y comida saludable http://ar.mujer.yahoo.com/cocina/

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directly mix 3 way voice
by Lee JJ 02 Sep '08

02 Sep '08

Hello : Yes , I am doing SIP. I would like to do something like call jump , while 2 leg talking NOT interrupt anyone leg , neither into music nor into silence. Someone can originate another endpoint phone ring , and switch over . It seems only use conference to achieve it. It seems hard to transfer an already bridged call into conference room, neither the originated call from scripts. # calltest.js n_sess = new Session() ; res = n_sess.originate(n_sess, "sofia/inter2/1527%210.243.126.72" ); uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 56089ae3-134c-4bfc-a697-b08194918bf2,2008-08-27 13:35:33,1219815333,sofia/inter2/1527,CS_SOFT_EXECUTE,FreeSWITCH,0000000000,,1527,,,,,PCMU,8000,PCMU,8000 >Didn't you say you're doing SIP? To not have it put into hold music >set the variable hold_music=silence and nobody will get music while >setting up the threeway call. >/b >>On Aug 28, 2008, at 2:46 AM, Lee JJ wrote: >> Hello : >> >> Is it possible to directly mix 3 way voice ? >> Not putting eny leg into holding music. >>Brian West

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Can I use the dialplan to execute specific tasks after an incoming all has been bridged?
by Jon Bruel 02 Sep '08

02 Sep '08

I wish to do some variable bookkeeping and database manipulation after an incoming call has been bridged - but before hangup. If possible, I would prefer to keep the dialplan lean and not to use the javascript application, but I havn't figured out how. In terms of Asterisk, I would like to emulate the dial M(macroname) option. What are the options for doing that? And are there special consideration with respect to accessing channel-A vs. channel-B variables?

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Query about javascript
by Ruchir Brahmbhatt 02 Sep '08

02 Sep '08

Hi all, Is it possible to call mysql stored procedure from javascript in freeswitch?

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Re: [Freeswitch-users] How to configure FreeSWITCH for callingSIPprofile "internal"?
by Евгений Золотов 02 Sep '08

02 Sep '08

On 1st host (192.168.2.107, it opos8, it 127.0.0.1) are carried out: - SIPP UAC - SIPP UAS - FS 1. At first we do direct connect RTP UAC-> UAS and return relaying UAS-> UAC [root@opos9 sipp.svn]# ./sipp -sn uac_pcap -m 1 opos9 [olej@opos9 sipp.svn]$ ./sipp -sn uas -rtp_echo [root@opos9 OGG]# tcpdump -i3 -n host opos9 tcpdump: WARNING: Promiscuous mode not supported on the "any" device tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on any, link-type LINUX_SLL (Linux cooked), capture size 96 bytes 11:24:58.171257 IP 127.0.0.1.sip-tls > 127.0.0.1.sip: SIP, length: 559 11:24:58.171514 IP 127.0.0.1.sip > 127.0.0.1.sip-tls: SIP, length: 301 11:24:58.171616 IP 127.0.0.1.sip > 127.0.0.1.sip-tls: SIP, length: 458 11:24:58.172038 IP 127.0.0.1.sip-tls > 127.0.0.1.sip: SIP, length: 351 11:24:58.172404 IP 127.0.0.1.6001 > 127.0.0.1.6000: UDP, length 252 11:24:58.172449 IP 127.0.0.1.6000 > 127.0.0.1.6001: UDP, length 252 11:24:58.204027 IP 127.0.0.1.6001 > 127.0.0.1.6000: UDP, length 252 11:24:58.204080 IP 127.0.0.1.6000 > 127.0.0.1.6001: UDP, length 252 .................... 11:25:06.314426 IP 127.0.0.1.6001 > 127.0.0.1.6000: UDP, length 16 11:25:06.314465 IP 127.0.0.1.6001 > 127.0.0.1.6000: UDP, length 16 11:25:06.314477 IP 127.0.0.1.6001 > 127.0.0.1.6000: UDP, length 16 11:25:06.314767 IP 127.0.0.1.6000 > 127.0.0.1.6001: UDP, length 16 11:25:06.314793 IP 127.0.0.1.6000 > 127.0.0.1.6001: UDP, length 16 11:25:06.314804 IP 127.0.0.1.6000 > 127.0.0.1.6001: UDP, length 16 11:25:07.174327 IP 127.0.0.1.sip-tls > 127.0.0.1.sip: SIP, length: 351 11:25:07.174526 IP 127.0.0.1.sip > 127.0.0.1.sip-tls: SIP, length: 293 498 packets captured 1494 packets received by filter 0 packets dropped by kernel - it's OK - RTP trafic run into 2 direction. 2. Now it would be desirable to include FreeSWITCH in stream rupture between UAC - UAS: [olej@opos9 sipp.svn]$ ./sipp -sn uas -rtp_echo -p 5062 -mp 6004 - RTP port 6004 [root@opos9 sipp.svn]# ./sipp -sn uac_pcap -s 20081 -m 1 -mp 6008 192.168.2.107:5070 - RTP port 6008 [root@opos9 OGG]# tcpdump -i3 -n dst host 192.168.2.107 or 127.0.0.1 and src host 192.168.2.107 or 127.0.0.1 tcpdump: WARNING: Promiscuous mode not supported on the "any" device tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on any, link-type LINUX_SLL (Linux cooked), capture size 96 bytes 15:27:06.474765 IP 192.168.2.107.sip-tls > 192.168.2.107.5070: UDP, length 563 15:27:06.475596 IP 192.168.2.107.5070 > 192.168.2.107.sip-tls: UDP, length 310 15:27:06.487313 IP 192.168.2.107.5070 > 127.0.0.1.5062: UDP, length 921 15:27:06.487707 IP 192.168.2.107.5062 > 192.168.2.107.5070: UDP, length 322 15:27:06.487859 IP 192.168.2.107.5062 > 192.168.2.107.5070: UDP, length 479 15:27:06.489679 IP 192.168.2.107.5070 > 127.0.0.1.5062: UDP, length 355 15:27:06.491973 IP 192.168.2.107.5070 > 192.168.2.107.sip-tls: UDP, length 590 15:27:06.493873 IP 192.168.2.107.5070 > 192.168.2.107.sip-tls: UDP, length 822 15:27:06.496382 IP 192.168.2.107.sip-tls > 192.168.2.107.5070: UDP, length 354 15:27:06.497854 IP 127.0.0.1.6008 > 127.0.0.1.0: UDP, length 252 15:27:06.497898 IP 127.0.0.1 > 127.0.0.1: ICMP 127.0.0.1 udp port 0 unreachable, length 288 15:27:06.529517 IP 127.0.0.1.6008 > 127.0.0.1.0: UDP, length 252 15:27:06.529558 IP 127.0.0.1 > 127.0.0.1: ICMP 127.0.0.1 udp port 0 unreachable, length 288 15:27:06.559627 IP 127.0.0.1.6008 > 127.0.0.1.0: UDP, length 252 .... 15:27:14.640995 IP 127.0.0.1.6008 > 127.0.0.1.0: UDP, length 16 15:27:14.641035 IP 127.0.0.1 > 127.0.0.1: ICMP 127.0.0.1 udp port 0 unreachable, length 52 15:27:14.641061 IP 127.0.0.1.6008 > 127.0.0.1.0: UDP, length 16 15:27:14.641071 IP 127.0.0.1 > 127.0.0.1: ICMP 127.0.0.1 udp port 0 unreachable, length 52 15:27:14.641086 IP 127.0.0.1.6008 > 127.0.0.1.0: UDP, length 16 15:27:14.641095 IP 127.0.0.1 > 127.0.0.1: ICMP 127.0.0.1 udp port 0 unreachable, length 52 15:27:15.501954 IP 192.168.2.107.sip-tls > 192.168.2.107.5070: UDP, length 354 15:27:15.502513 IP 192.168.2.107.5070 > 192.168.2.107.sip-tls: UDP, length 485 15:27:15.511937 IP 192.168.2.107.5070 > 127.0.0.1.5062: UDP, length 615 15:27:15.513052 IP 192.168.2.107.5062 > 192.168.2.107.5070: UDP, length 314 505 packets captured 1516 packets received by filter 0 packets dropped by kernel <extension name="20081"> <condition field="destination_number" expression="^20081$" > <action application="set" data="bypass_media=true" /> <action application="bridge" data="sofia/nat/20081@127.0.0.1:5062" /> <action application="answer" /> <action application="sleep" data="1000" /> </condition> </extension> It is well visible that SIP-stream through FreeSWITCH is well transmit between UAs. But RTP-stream goes to FreeSWITCH, but it does not reach. What is it? We do something not so? Or it is misoperation of FreeSWITCH? This is a prtocol from FreeSWITCH console: 2008-09-02 15:27:06 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing sipp->20081 in context public 2008-09-02 15:27:06 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/nat/20081@127.0.0.1:5062 [b808d845-f4e6-41e1-82d8-1251c0aec269] 2008-09-02 15:27:06 [NOTICE] sofia.c:2195 sofia_handle_sip_i_state() Ring-Ready sofia/nat/20081@127.0.0.1:5062! 2008-09-02 15:27:06 [NOTICE] mod_sofia.c:1071 sofia_receive_message() Ring-Ready sofia/nat/sipp@127.0.0.1:5061! 2008-09-02 15:27:06 [NOTICE] sofia.c:2484 sofia_handle_sip_i_state() Channel [sofia/nat/20081@127.0.0.1:5062] has been answered 2008-09-02 15:27:06 [NOTICE] sofia.c:2497 sofia_handle_sip_i_state() Channel [sofia/nat/sipp@127.0.0.1:5061] has been answered 2008-09-02 15:27:15 [NOTICE] sofia.c:2573 sofia_handle_sip_i_state() Hangup sofia/nat/sipp@127.0.0.1:5061 [CS_HIBERNATE] [NORMAL_CLEARING] 2008-09-02 15:27:15 [NOTICE] switch_ivr_bridge.c:594 signal_bridge_on_hangup() Hangup sofia/nat/20081@127.0.0.1:5062 [CS_HIBERNATE] [NORMAL_CLEARING] 2008-09-02 15:27:15 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 3 (sofia/nat/sipp@127.0.0.1:5061) Ended 2008-09-02 15:27:15 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/nat/sipp@127.0.0.1:5061 [CS_HANGUP] 2008-09-02 15:27:15 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 4 (sofia/nat/20081@127.0.0.1:5062) Ended 2008-09-02 15:27:15 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/nat/20081@127.0.0.1:5062 [CS_HANGUP] - FreeSWITCH "thinks", that all is fine? :(

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How to configure FreeSWITCH for calling SIP profile "internal"?
by Евгений Золотов 02 Sep '08

02 Sep '08

How I should reconstruct configuration files of FreeSWITCH, that any user could carry out call at SIP profile "internal" port 5060 and also it was not required to its registration (message REGISTER from one)? Like at profiles "external" and "nat" ( version 1.0.trunk( 9377 ) - earlier these profiles were called in another way: "default" and "outbound"?). ) It's necessary for tests - calls from SIPP client's side, which in standard scenarios does not cause REGISTER and carries out INVITE with a name "sipp". With gratitude, Evgeniy.

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