- FreeSWITCH-users - FreeSWITCH Mailing Lists

Call-Blocking application
by Alois Komenda 15 Jul '08

15 Jul '08

Hello, how can I realize a Call-Blocking application with FreeSWITCH? The application should look up the caller in a kind of blacklist or delegate the lookup to another application. Every incoming call has to be blocked if the caller is on the blacklist. Is it possible to realize this as a application which is called from the dialplan? As the dialplan is parsed only once for a call, it won't work if the application is called from an action tag, I guess. The outcome of the application can't influence any following action, right? What about calling the application in the condition tag? Would that count as an "abuse" as mentioned in the wiki? What method can you recommend to block incoming calls depending on the originator? Regards, -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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outbound fxo line pooling
by John Wehle 15 Jul '08

15 Jul '08

I currently have: <extension name="outgoing-fxo"> <condition field="destination_number" expression="^(4[0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> <action application="bridge" data="openzap/1/1/${dialed_ext}"/> </condition> </extension> which routes calls to extension 4XX out the first openzap line. How do I set up a pool of openzap lines and route calls to extension 4XX to any available openzap line? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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send_dtmf problems
by John Wehle 15 Jul '08

15 Jul '08

I need to send some dtmf tones after handling vmail for the System 25 so I've set up (for testing purposes) an extension which should just answer the phone and play dtmf: <extension name="system25_vmail"> <condition field="destination_number" expression="^(5590)|(55[1-8])$"> <action application="answer"/> <action application="send_dtmf" data="1234567890@500"/> <action application="sleep" data="5000"/>  </condition> </extension> 1) When I dial the extension from a Grandstream GXP-2000 a network trace shows the digits being sent from FreeSWITCH as RTP events, however no dtmf audio tones are sent. How do I configure FreeSWITCH to send both RTP digits and inband audio? 2) When I dial the extension from a FXO port on a Sangoma A204D it connects, after five seconds it sends two tones, and then disconnects. How do I configure FreeSWITCH to send all the dtmf tones? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Problem: call routing via external gateway not working
by Birgit Arkesteijn 14 Jul '08

14 Jul '08

Hi, I'm running FreeSWITCH Version 1.0.trunk (498:8901) on Suse 10.0 (x86_64). I'm a novice in setting up the configuration, so sorry if this is a trivial one. I'm trying to get the following test setup to work: 1. SIP desktop phone 2. -> Asterisk on risk.westhawk.co.uk 3. -> FreeSWITCH on apso.westhawk.co.uk (using xml_curl to get a dialplan) 4. -> Asterisk on risk.westhawk.co.uk 5. -> another SIP desktop phone I'm fine up to (and including) step 3: - I can see the incoming call - I can see the dialplan being accessed - I can see the dialplan being matched - I can see from the freeswitch log it's trying to bridge I'm stuck going to step 4: - I cannot see any SIP packets going out back to risk We checked our firewall log files, but no packets are dropped to risk (as far as we can tell). What I've configured is: - a user in directory/ so the incoming call works - an external gateway, called 'doneright', for the outgoing leg Below are - snippets from the freeswitch.log file, - some 'sofia status' commands on the console, - the content of my doneright.xml external gateway file The 'doneright' gateway doesn't currently register at risk, because we don't think we need to. When we omitted the 'register' setting by accident in the doneright.xml file, we noticed sip registration requests coming in at risk. I'm very puzzled by this all. On a different machine, we've done SIP routing to an external (then called 'outbound') SIP provider without a problem. Does anyone have a clue or some pointers how I can approach this problem? Thanks, Birgit *** Freeswitch log file: * (incoming call): 2008-07-09 11:36:00 [NOTICE] sofia.c:1514 config_sofia() Adding Alias [apso.westhawk.co.uk] for profile [internal] 2008-07-09 11:46:24 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/apso(a)192.67.4.83[466eaeee-4da4-11dd-b14f-7bc0d4e408b2] 2008-07-09 11:46:24 [DEBUG] sofia.c:1748 sofia_handle_sip_i_state() Channel sofia/internal/apso(a)192.67.4.83 entering state [received] * (dialplan): 2008-07-09 11:35:34 [NOTICE] mod_xml_curl.c:300 do_config() Binding [doneright-voipuri] XML Fetch Function [http://apso.westhawk.co.uk/test_dialplan/static_dialplan.xml][dialplan] 2008-07-09 11:46:24 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Birgit Arkesteijn->801@default 2008-07-09 11:46:24 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(801) =~ /^801$/ * (bridge): 2008-07-09 11:46:24 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/internal/apso(a)192.67.4.83 Execute bridge(sofia/gateway/doneright/1234) * (external gateway): 2008-07-09 11:46:24 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/external/1234 [467a1040-4da4-11dd-b14f-7bc0d4e408b2] 2008-07-09 11:46:24 [DEBUG] mod_sofia.c:1870 sofia_outgoing_channel() sofia/external/1234 State Change CS_NEW -> CS_INIT 2008-07-09 11:46:24 [DEBUG] switch_core_session.c:720 switch_core_session_signal_state_change() Kill sofia/external/1234 [BREAK] 2008-07-09 11:46:24 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/1234 Running State Change CS_INIT 2008-07-09 11:46:24 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (sofia/external/1234) State INIT 2008-07-09 11:46:24 [DEBUG] mod_sofia.c:80 sofia_on_init() sofia/external/1234 SOFIA INIT 2008-07-09 11:46:44 [DEBUG] sofia.c:194 sofia_event_callback() event [nua_i_state] status [487][Request Terminated] session: sofia/internal/apso(a)192.67.4.83 *** Console: > sofia status API CALL [sofia(status)] output: Name Type Data State ========================================================================== external profile sip:mod_sofia@192.67.4.56:5080 RUNNING (0) internal profile sip:mod_sofia@192.67.4.56:5060 RUNNING (0) nat profile sip:mod_sofia@192.67.4.56:5070 RUNNING (0) doneright gateway sip:apso@risk.westhawk.co.uk NOREG default alias internal ALIASED apso.westhawk.co.uk alias internal ALIASED outbound alias external ALIASED ========================================================================== 3 profiles 3 aliases > sofia status profile external API CALL [sofia(status profile external)] output: ========================================================================== Name external Domain Name external DBName sofia_reg_external Dialplan XML RTP-IP 192.67.4.56 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.67.4.56 Ext-SIP-IP 192.67.4.56 URL sip:mod_sofia@192.67.4.56:5080 BIND-URL sip:mod_sofia@192.67.4.56:5080;maddr=192.67.4.56 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ========================================================================== ========================================================================== > sofia status gateway doneright API CALL [sofia(status gateway doneright)] output: ========================================================================== Name doneright Scheme Digest Realm risk.westhawk.co.uk Username apso Password yes From <sip:apso@risk.westhawk.co.uk;transport=udp> Contact <sip:apso@192.67.4.56:5080;transport=udp> To sip:apso@risk.westhawk.co.uk Proxy sip:risk.westhawk.co.uk Context default Expires 600 Freq 598 Ping 0 PingFreq 0 State NOREG Status UP ========================================================================== *** /usr/local/freeswitch/conf/sip_profiles/external/doneright.xml: <include> <gateway name="doneright"> <param name="username" value="apso"/> <param name="password" value="xxxxxxx"/> <param name="realm" value="risk.westhawk.co.uk"/> <param name="proxy" value="risk.westhawk.co.uk"/> <param name="register" value="false"/> <param name="expire-seconds" value="600"/> </gateway> </include> -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Freeswitch on OpenBSD
by Mucker 14 Jul '08

14 Jul '08

Hi All, Anybody else trying to get Freeswitch up and running on OpenBSD? Following the install instructions for FreeBSD "seems" to work. But, when I start freeswitch I see a critical error: 2008-07-14 16:56:02 [CRIT] switch_loadable_module.c:751 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **File not found** # ls -al /usr/local/freeswitch/mod/ | grep sofia -rw-r--r-- 1 root wheel 7907294 Jul 14 11:02 mod_sofia.a -rw-r--r-- 1 root wheel 1142 Jul 14 11:02 mod_sofia.la #

3 6

acl list not correct?
by Ivan C Myrvold 14 Jul '08

14 Jul '08

I have the list below added at the end of acl.conf.xml, and incoming calls from 81.201.82.23, 81.201.82.25, 81.201.82.26, 81.201.82.27 gets dropped, because FreeSWITCH sends a "SIP/2.0 407 Proxy Authentication Required". But calls coming in from 81.201.82.24 and 81.201.82.28 are always successful. The range below is what Voxbone have recommended to use. Ivan <list name="voxbone" default="deny"> <node type="allow" cidr="81.201.82.0/26"/> <node type="allow" cidr="81.201.82.128/26"/> <node type="allow" cidr="81.201.83.0/26"/> <node type="allow" cidr="81.201.84.0/26"/> <node type="allow" cidr="81.201.84.128/26"/> <node type="allow" cidr="81.201.86.0/26"/> </list>

4 9

Start Up Problem
by Brian Young 14 Jul '08

14 Jul '08

When I try to start freeswitch, I get this message [root@bigblue bin]# ./freeswitch Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml Cannot Initialize [[error near line 1245]: unexpected closing tag </gateways>] I've rebuild it twice now. Any help would be greatly appreciated. Thanks

2 1

Question about IAX?
by Brian West 14 Jul '08

14 Jul '08

Just doing a quick poll to see who would like to see IAX in FreeSWITCH with all the bells and whistles? Brian West sip:brian@freeswitch.org

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Trouble with SIP not found by name error
by Ryan WIlson 14 Jul '08

14 Jul '08

Hi, I'm getting the following error when I try to dial out a call to a SIP gateway. tport_by_addrinfo(0xecded60): not found by name */ 208.68.17.144:5060 I've googled the error but didn't come up with any answers. Hope someone can help! Thank you, Ryan soa_generate_offer(static::0xee2fc80, 0) called soa_static_offer_answer_action(0xee2fc80, soa_generate_offer): called soa_static(0xee2fc80, soa_generate_offer): generating local description soa_static(0xee2fc80, soa_generate_offer): upgrade with local description soa_sdp_mode_set(0x4177ba30, (nil), ""): called soa_static(0xee2fc80, soa_generate_offer): storing local description soa_get_local_sdp(static::0xee2fc80, [(nil)], [0x4177bb50], [0x4177bb4c]) called nta: selecting scheme sip sres_cache_get(0xeca0be0, NAPTR, "eico.thinktel.ca.") called rr found in cache: eico.thinktel.ca. 35 sres_cache_get(0xeca0be0, NAPTR, "eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "eico.thinktel.ca" NAPTR (cached) sres_cache_get(0xeca0be0, SRV, "_sip._udp.eico.thinktel.ca.") called rr found in cache: _sip._udp.eico.thinktel.ca. 33 sres_cache_get(0xeca0be0, SRV, "_sip._udp.eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "_sip._udp.eico.thinktel.ca" SRV (cached) sres_cache_get(0xeca0be0, SRV, "_sip._tcp.eico.thinktel.ca.") called rr found in cache: _sip._tcp.eico.thinktel.ca. 33 sres_cache_get(0xeca0be0, SRV, "_sip._tcp.eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "_sip._tcp.eico.thinktel.ca" SRV (cached) sres_cache_get(0xeca0be0, A, "eico.thinktel.ca.") called rr found in cache: eico.thinktel.ca. 01 sres_cache_get(0xeca0be0, A, "eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "eico.thinktel.ca" A (cached) nta: eico.thinktel.ca. IN A 208.68.17.144 tport_tsend(0xecded60) tpn = */208.68.17.144:5060 tport_resolve addrinfo = 208.68.17.144:5060 tport_by_addrinfo(0xecded60): not found by name */208.68.17.144:5060 tport_vsend returned 1042 nta: sent INVITE (101882567) to */208.68.17.144:5060 tport_pend(0xecded60): pending 0xee25ab0 for udp/10.0.4.5:5060 (already 0) nua(0xed0d980): call state changed: init -> calling, sent offer soa_get_local_sdp(static::0xee2fc80, [0x4177bb88], [0x4177bb80], [(nil)]) called nua: nua_application_event: entering nua(0xed0d980): sent signal r_invite nua: nua_handle_magic: entering nta: timer set next to 341 ms nta: timer A fired, retransmit INVITE (101882567) tport_tsend(0xecded60) tpn = */208.68.17.144:5060 tport_resolve addrinfo = 208.68.17.144:5060 tport_by_addrinfo(0xecded60): not found by name */208.68.17.144:5060 tport_vsend returned 1042 nta: resent INVITE (101882567) to */208.68.17.144:5060 tport_release(0xecded60): 0xee25ab0 by 0xee5e3f0 with (nil) tport_pend(0xecded60): pending 0xee25ab0 for udp/10.0.4.5:5060 (already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free nta: timer set next to 1000 ms nta: timer A fired, retransmit INVITE (101882567) tport_tsend(0xecded60) tpn = */208.68.17.144:5060 tport_resolve addrinfo = 208.68.17.144:5060 tport_by_addrinfo(0xecded60): not found by name */208.68.17.144:5060 tport_vsend returned 1042 nta: resent INVITE (101882567) to */208.68.17.144:5060 tport_release(0xecded60): 0xee25ab0 by 0xee5e3f0 with (nil) tport_pend(0xecded60): pending 0xee25ab0 for udp/10.0.4.5:5060 (already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free nta: timer set next to 2000 ms tport_wakeup_pri(0xecded60): events IN tport_recv_event(0xecded60) tport_recv_iovec(0xecded60) msg 0xee30a20 from (udp/10.0.4.5:5060) has 4 bytes, veclen = 1 tport_deliver(0xecded60): bad msg 0xee30a20 (4 bytes) from udp/ 10.10.0.19:5060/sip next=(nil) nta: timer A fired, retransmit INVITE (101882567) tport_tsend(0xecded60) tpn = */208.68.17.144:5060 here is the config 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <?xml version="1.0"?> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <document type="freeswitch/xml"> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <section name="configuration" description="Various Configuration"> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <configuration name='sofia.conf' description='sofia endpoint'> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <profiles> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <profile name='onepipe'> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <settings> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='nonce-ttl' value='60'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='use-rtp-timer' value='true'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='ext-sip-ip' value='64.69.80.34'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='rtp-timer-name' value='soft'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='codec-prefs' value='g729'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='debug' value='1'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='rfc2833-pt' value='101'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='ext-rtp-ip' value='64.69.80.34'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='dtmf-duration' value='100'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='sip-port' value='5060'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='dialplan' value='XML,enum'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='rtp-ip' value='10.0.4.5'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='codec-ms' value='20'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='sip-ip' value='10.0.4.5'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='accept-blind-reg' value='1'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </settings> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <gateways> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <gateway name='SBC'> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='username' value='6046309907'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='retry-seconds' value='30'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='realm' value='eico.thinktel.ca'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='extension' value='6046309907'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='caller-id-in-from' value='0'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='register' value='0'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='from-domain' value='eico.thinktel.ca'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='proxy' value='eico.thinktel.ca'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='password' value='6046309907'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='expire-seconds' value='60'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </gateway> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </gateways> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </profile> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </profiles> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </configuration> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </section> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </document> 2008-07-14 02:14:15+ Ryan Wilson OnePipe Communications Inc. t. 778.786.2222 c. 604.716.2222 f. 866.791.7120 www.onepipe.ca

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FreeSWITCH and NAT problem
by Diego Viola 14 Jul '08

14 Jul '08

Hi, I'm running FreeSWITCH on a server that has no NAT in between... the server is on Slicehost (slicehost.com) a VPS. I'm trying to register my softphone (Zoiper and X-Lite) from my computer, my computer is behind a router (NAT). So the scenario is the following: Softphone (Zoiper or X-Lite) -> router (NAT) -> INTERNET -> FS server (no nat -- public ip address). The problem is that I can't make it register, sometimes it will register and sometimes it wont, this problem is randomly. This is what I have in vars.xml: <X-PRE-PROCESS cmd="set" data="domain=maitei.com.py"/> I'm not sure if this has something to be with STUN or NAT, it works fine with Asterisk... in Asterisk I have nat=yes in my SIP user (sip.conf), is there a equivalent of nat=yes in FS that I can use for my user? Thanks, Diego

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