October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

problem with group call
by Jason Holden 13 Nov '13

13 Nov '13

I currently have a freeswitch running 1.3.5 and am having the following problem. When bridging to a group only the first extension in the directory is rining. I've tried added the +A flag with no success. Any suggestions on how I can have all pointers in the group call ring?

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blind transfer back to caller on failure
by Tiago Sampaio 11 Nov '13

11 Nov '13

Hi all. I'm wondering if is possible to return the call if no answer or user busy? i tryed this: <context name="INTERNAL"> <extension name="INTER_RAMAL"> <condition field="destination_number" expression="^(\d{4})$"> <action application="set" data="continue_on_fail=true" /> <action application="bridge" data="[originate_timeout=10]user/$1" /> </condition> </extension> <extension name="BLIND_TRANSFER"> <condition field="destination_number" expression="^blind_transfer$"> <action application="answer"/> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 5000 #"/> <action application="set" data="continue_on_fail=true" /> <action application="execute_extension" data="is_transfer XML INTERNAL" />  <action application="bridge" data="user/${caller_id_number}" /> <action application="hangup" /> </condition> </extension> <extension name="is_transfer"> <condition field="destination_number" expression="^is_transfer$"/> <condition field="${digits}" expression="^(\d+)$"> <action application="set" data="continue_on_fail=true" /> <action application="transfer" data="-bleg ${digits} XML INTERNAL"/> <anti-action application="eval" data="cancel transfer"/> </condition> </extension> </context> Example: User A call user B, user B answer and do blind transfer to user C. User B hangup when call blind transfer and C rings 30 seconds and then transfer again to B before hangup A... Now if i do blind transfer and no answer or user busy, freeswitch hangup all sides...

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FS with SSL/TLS issues!
by mehroz 10 Nov '13

10 Nov '13

Hi, it seems like i am unable to get it done. Here is what i have done 1. created CA (Root) certificates: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org 2. created Server Certificate: ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org *I replaced pbx.freeswitch.org and freeswitch.org with my FS IP address in which i am listening.* 3. Enabled /internal_ssl_enable=true/ and/ ip_tls_version=tlsv1/ in vars.xml . 4. When i try to Register 3CX on TLS over port 5061, all i see is : tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0x7fbd94e7a5e0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2084 tport_close() tport_close(0x7fbd94e7a5e0): tls/59.103.76.82:56863/sips tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fbd94005090): events IN tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fbd94005090): new secondary tport 0x7fbd94f53c70 tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fbd94f53c70): new connection from tls/59.103.76.82:56864/sips. PLease help!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Is it necessary to call hangup....
by Moiz Chinoy 07 Nov '13

07 Nov '13

Hello, Is it necessary to call hangup in dialplan after say playing a file through playback application. -- Regards, Moiz Chinoy.

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481 Call Does Not Exist
by Miha 04 Nov '13

04 Nov '13

Hi, I need a little help with figuring out why FS sends "481". This is happing when call has been forward (302). All calls are relayed throught opensips as I am using it for registrations and load_balancing. http://pastebin.freeswitch.org/21555 tnx! miha

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Raspberry Pi, mod_gsmopen GSM dongle
by Sergey Zhuravlov 04 Nov '13

04 Nov '13

Hi! Does somebody have a positive experience with Raspberry Pi, mod_gsmopen GSM dongle ??? I use Raspberry Pi B and e1550. It's a shame that with an asterisk all works fine -- RasPBX. http://www.raspberry-asterisk.org/ But I prefer the FreeSWITCH ;-) On the basis of experience, I thought that PS will work best for this modest hardware faster and require fewer resources. But it turns out it is not. Asterisk with a web server, MySQL database, and other, less load than FS. In this connection, other parameters may be used starting FS to save resources? And the main question! Silence an incoming call on 5000 (IVR) as well as an outgoing in gsm network. Those are not sound. -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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ACK disappeared from SIP session, Help
by "Гришин В.С." 01 Nov '13

01 Nov '13

the scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone FreeSWITCH IP is 191.21.19.12 Asterisk IP is 162.48.13.26 Call comes to FreeSWITCH from a Cisco 2811. SIP session between freeswitch and asterisk(1.8.20) |Time | 191.21.19.12 | | | | 162.48.13.26 | |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.065794| 100 Trying| |SIP Status | |(5065) <------------------ (5060) | |21.122406| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |21.143580| RTP (g711A) |RTP Num packets:50 Duration:0.922s SSRC:0xB5320A6 | |(21866) <------------------ (12772) | |22.282684| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |22.296762| 181 Call is being forwarded |SIP Status | |(5065) <------------------ (5060) | |27.558268| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |35.039883| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.307222| RTP (g711A) |RTP Num packets:1490 Duration:29.773s SSRC:0x530CAB37 | |(21866) <------------------ (12772) | |35.476614| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.532995| RTP (g711A) |RTP Num packets:2966 Duration:29.680s SSRC:0x308FC816 | |(21866) ------------------> (12772) | |36.436846| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |38.282686| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |42.025399| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |45.637280| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |49.343115| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |53.119940| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |56.907800| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |60.791453| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |64.606730| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |65.219392| BYE | |SIP Request | |(5065) <------------------ (5060) | |65.234354| 200 OK | |SIP Status | |(5065) ------------------> (5060) | |65.234359| 200 OK | |SIP Status | |(5065) ------------------> (5060) | It is visible that there is no package with ACK confirmation from a FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a freeswitch, and without having waited confirmations (ACK) after 23 seconds sends SIP BY on a freeswitch and on IP Phone. After that call stops. Before disconnection subscribers hear each other of 29 (+-) seconds In tcpdump between Cisco and FreeSWITCH it is visible that the SIP ACK packet from Cisco arrives on a freeswitch. In what place the SIP ACK in a FreeSWITCH disappeared? What to me to analyse in SIP session? Maybe someone solved a similar problem? please, Help! Vladislav Grishin

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mod_lcr and UnixODBC Error
by FERNANDO VILLARROEL 31 Oct '13

31 Oct '13

Dear all. I am trying to use mod_lcr, but in my tests i did received the follow error: 2013-10-31 16:35:03.916887 [ERR] switch_core_sqldb.c:1114 ERR: [SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND ( (digits IN (12145551111, 1214555111, 121455511, 12145551, 1214555, 121455, 12145, 1214, 121, 12, 1) AND lrn = false) OR (digits IN (12145551111, 1214555111, 121455511, 12145551, 1214555, 121455, 12145, 1214, 121, 12, 1) AND lrn = true)) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, rate, quality DESC, reliability DESC, random();] [STATE: 42883 CODE 7 ERROR: [unixODBC]ERROR: el operador no existe: prefix_range = bigint; Error while executing the query ] I am using PostgreSQL 8.4.11 and Debian Squezee How i can solve this issue? Regards

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bind_digit_action stops working when the call is sent to a socket
by Ali Pey 31 Oct '13

31 Oct '13

Hello, As soon as the call is sent to a socket, bind digit actions stop working. <action application="socket" data="127.0.0.1:9090 full"/> I have tried setting bind_digit_action both before and after the call is sent to the socket. It doesn't work. Before call is sent to socket, I see: 2013-10-31 10:48:38.183166 [DEBUG] switch_rtp.c:3915 RTP RECV DTMF 5:3520 2013-10-31 10:48:38.183166 [DEBUG] mod_dptools.c:186 sofia/sip-trunks/ 6132952508(a)qasip.onebox.com Digit match binding [exec:playback][ivr/ivr-please_enter_extension_followed_by_pound.wav] After is sent to the socket, it becomes: 2013-10-31 11:07:38.042566 [DEBUG] switch_rtp.c:3915 RTP RECV DTMF 5:4160 2013-10-31 11:07:38.042566 [DEBUG] switch_channel.c:419 RECV DTMF 5:4160 2013-10-31 11:07:38.042566 [DEBUG] mod_dptools.c:2049 Digit 5 Any ideas? Is this a bug? Is there a work around? Thanks, Ali Pey

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max-sessions value small may cause crash
by Ashou Han 31 Oct '13

31 Oct '13

Hi, My FS is FreeSWITCH Version 1.2.12+git~20130910T181609Z~f5ba0bb7d4 (git f5ba0bb 2013-09-10 18:16:09Z) I set <param name="max-sessions" value="20"/>, <param name="sessions-per-second" value="10"/> in switch.conf.xml FS carshed very offen like this in messages log: Oct 26 19:10:21 fs kernel: [4592903.781042] freeswitch[29353]: segfault at 0 ip 00007f4e22e0c8cd sp 00007f4e0924e5b0 error 4 in libfreeswitch.so.1.0.0[7f4e22ce4000+229000] Oct 26 23:41:45 fs kernel: [4609190.441226] freeswitch[518]: segfault at 0 ip 00007ff71baf58cd sp 00007ff7068ae5b0 error 4 in libfreeswitch.so.1.0.0[7ff71b9cd000+229000] Oct 28 09:57:58 fs kernel: [89606.041406] freeswitch[23780]: segfault at 0 ip 00007f68651d78cd sp 00007f684ff205b0 error 4 in libfreeswitch.so.1.0.0[7f68650af000+229000] Oct 29 16:55:24 fs kernel: [201076.501294] freeswitch[16422]: segfault at 0 ip 00007f0b68fc48cd sp 00007f0b4f50d5b0 error 4 in libfreeswitch.so.1.0.0[7f0b68e9c000+229000] lots in freeswitch.log: 2013-10-25 15:25:24.651750 [CRIT] switch_core_session.c:2228 Over Session Limit! 20 2013-10-25 15:25:24.651750 [CRIT] mod_sofia.c:4869 Error Creating Session 2013-10-25 18:36:03.311729 [CRIT] switch_core_session.c:2228 Over Session Limit! 20 2013-10-25 18:36:03.311729 [CRIT] mod_sofia.c:4869 Error Creating Session ... 2013-10-30 15:01:46.501223 [CRIT] mod_sofia.c:4869 Error Creating Session 2013-10-30 15:01:46.521224 [CRIT] switch_time.c:1007 Over Session Rate of 10! May this cause the FS crash? I have another FS server with same Hardware and FS version, but <param name="max-sessions" value="2000"/> and no crash issues. Thanks, Ashou Han

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