- FreeSWITCH-users - FreeSWITCH Mailing Lists

Calls Authentication
by Lloyd Aloysius 27 Feb '14

27 Feb '14

Hello All, I have the the following set in my profile parameters. challenge-realm = false auth-all-packets = true But someone from the otuside can enter into my dial plan. Why the switch is not blocking(authenticate) before enter into the dialpaln. I do not have any default context or user 1001 in the system. please see the log below. What i am doing wrong in my setup. ========================= 2014-02-27 07:07:16.449120 [DEBUG] sofia.c:7868 sofia/sipinterface_1/1001@{freeswitch_ip} receiving invite from {originate_ip}:10036 version: 1.2.22 git aabf813 2014-02-24 01:22:36Z 64bit 2014-02-27 07:07:16.449120 [DEBUG] sofia.c:8025 IP {originate_ip} Rejected by acl "domains". Falling back to Digest auth. *2014-02-27 07:07:16.449120 [WARNING] sofia_reg.c:1532 SIP auth challenge (INVITE) on sofia profile 'sipinterface_1' for [018584290411@{freeswitch_ip}] from ip {originate_ip}* 2014-02-27 07:07:16.449120 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/1001@{freeswitch_ip}) State NEW 2014-02-27 07:07:16.449120 [DEBUG] switch_core_session.c:1016 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.449120 [DEBUG] sofia.c:1792 detaching session b31c0e64-9fa7-11e3-8ba7-6d0b1a3b8605 2014-02-27 07:07:16.809118 [DEBUG] sofia.c:1884 Re-attaching to session b31c0e64-9fa7-11e3-8ba7-6d0b1a3b8605 2014-02-27 07:07:16.809118 [DEBUG] switch_core_session.c:1016 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.809118 [DEBUG] switch_core_session.c:1016 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.809118 [DEBUG] sofia.c:7868 sofia/sipinterface_1/1001@{freeswitch_ip} receiving invite from {originate_ip}:10036 version: 1.2.22 git aabf813 2014-02-24 01:22:36Z 64bit *2014-02-27 07:07:16.809118 [DEBUG] sofia.c:8025 IP {originate_ip} Rejected by acl "domains". Falling back to Digest auth.* 2014-02-27 07:07:16.829310 [DEBUG] sofia.c:5811 Channel sofia/sipinterface_1/1001@{freeswitch_ip} entering state [received][100] 2014-02-27 07:07:16.829310 [DEBUG] sofia.c:5824 Remote SDP: v=0 o=- 5753281 5753289 IN IP4 {originate_ip} s=eyeBeam c=IN IP4 192.168.1.100 t=0 0 m=audio 11424 RTP/AVP 100 6 0 8 3 18 97 5 101 a=rtpmap:100 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : 86FF30D0 0000006D 192.168.1.100 24006 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[G722:9:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[PCMU:0:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[PCMA:8:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [speex:100:16000:20:0]/[GSM:3:8000:20:13200] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[G722:9:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[PCMU:0:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[PCMA:8:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [DVI4:6:16000:20:0]/[GSM:3:8000:20:13200] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:3190 Set Codec sofia/sipinterface_1/1001@{freeswitch_ip} PCMU/8000 20 ms 160 samples 64000 bits 2014-02-27 07:07:16.829310 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/1001@{freeswitch_ip} Original read codec set to PCMU:0 2014-02-27 07:07:16.829310 [DEBUG] sofia_glue.c:5451 Set 2833 dtmf send/recv payload to 101 2014-02-27 07:07:16.829310 [DEBUG] sofia.c:6091 (sofia/sipinterface_1/1001@{freeswitch_ip}) State Change CS_NEW -> CS_INIT 2014-02-27 07:07:16.829310 [DEBUG] switch_core_session.c:1351 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_1/1001@{freeswitch_ip}) Running State Change CS_INIT 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:455 (sofia/sipinterface_1/1001@{freeswitch_ip}) State INIT 2014-02-27 07:07:16.829310 [DEBUG] mod_sofia.c:87 sofia/sipinterface_1/1001@{freeswitch_ip} SOFIA INIT 2014-02-27 07:07:16.829310 [DEBUG] mod_sofia.c:127 (sofia/sipinterface_1/1001@{freeswitch_ip}) State Change CS_INIT -> CS_ROUTING 2014-02-27 07:07:16.829310 [DEBUG] switch_core_session.c:1351 Send signal sofia/sipinterface_1/1001@{freeswitch_ip} [BREAK] 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:455 (sofia/sipinterface_1/1001@{freeswitch_ip}) State INIT going to sleep 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_1/1001@{freeswitch_ip}) Running State Change CS_ROUTING 2014-02-27 07:07:16.829310 [DEBUG] switch_channel.c:2163 (sofia/sipinterface_1/1001@{freeswitch_ip}) Callstate Change DOWN -> RINGING 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:471 (sofia/sipinterface_1/1001@{freeswitch_ip}) State ROUTING 2014-02-27 07:07:16.829310 [DEBUG] mod_sofia.c:150 sofia/sipinterface_1/1001@{freeswitch_ip} SOFIA ROUTING 2014-02-27 07:07:16.829310 [DEBUG] switch_core_state_machine.c:117 sofia/sipinterface_1/1001@{freeswitch_ip} Standard ROUTING *2014-02-27 07:07:16.829310 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->018584290411 in context default* Thanks and regards, Lloud

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Jira has been upgraded.
by Ken Rice 27 Feb '14

27 Feb '14

Hey Guys, Jira.freeswitch.org has been upgraded. Let me know if you run across any issues with it. You¹ll notice quite a few changes. We¹re also upgrading and updating a few other systems. So watch out for brief outages on other systems. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

4 7

fax_header & fax_ident length
by Tihomir Culjaga 26 Feb '14

26 Feb '14

hello every1 just a small question ... what is the max length of fax_header and fax_ident. from spandsp source i have found its 50 characters for fax_header and 22 for ident... but im having issues going over 26 characters for header. The fax simply doesn't go through... so im wondering if there is a limit on the T.30 protocol for the header. anyone ?

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Significant audio clipping when bridging webrtc to sip
by Joegen Baclor 26 Feb '14

26 Feb '14

Hi, I am using FreeSwitch 1.4-beta to bridge WebRTC->SIP. All is working great except for the fact that audio is almost always clipped at the beginning of the call. Looking at the logs, it seems to me that this is due to the delay incurred when performing the DTLS handshake on receipt of media. I am doing a "pre_answer" before bridging. My question is, is it possible to do the DTLS handshake prior to originate and right after pre_answer? Log is in pastebin: http://pastebin.freeswitch.org/22037. The dropped media happens during the gap between lines 310 and 311. Thanks for any pointers. Joegen

2 2

ClueCon Weekly! Feb 26th Join us!
by Ken Rice 26 Feb '14

26 Feb '14

Hey FreeSWITCHers, Join us for ClueCon Weekly today at 1PM EST (10AM PST) at sip:888@conference.freeswitch.org Todays Topic with be FreeSWITCH Infrastructure Changes Update and Following up on BSDs plus Jira! Anthony, Brian and MikeJ along with several others have been busting hump this week. Join us to find out been going on here! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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http://jira.freeswitch.org/browse/FS-6227
by Brian West 26 Feb '14

26 Feb '14

FreeSWITCHers, Here is another example of a platform we’ve supported in the past thats shriveled up an died on the vine. Anyone interested in helping reviving support for Solaris/SmartOS? -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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NORMAL_UNSPECIFIED Disconnect
by Brian Fertig 26 Feb '14

26 Feb '14

List, I am having an interesting issue. I have active calls disconnecting at different points and it appears on the ladder graph that FS is sending BYE's to both legs of the call. I have looked at the log and cant make heads or tails as to why. I have included a snippet of the log in this email maybe someone can tell me whats going on. TIA! 2014-02-25 11:00:04.816713 [DEBUG] mod_sofia.c:87 sofia/internal/517NXXXXXX(a)216.238.XXX.XXX SOFIA INIT 2014-02-25 11:00:04.816713 [DEBUG] sofia_glue.c:2744 sofia/internal/517NXXXXXX(a)216.238.XXX.XXX sending invite version: 1.2.22 git f7d5eba 2014-02-18 21:16:05Z 64bit Local SDP: v=0 o=FreeSWITCH 1393320114 1393320115 IN IP4 144.202.XXX.XXX s=FreeSWITCH c=IN IP4 144.202.XXX.XXX t=0 0 m=audio 23890 RTP/AVP 18 0 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2014-02-25 11:00:04.836881 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/517NXXXXXX(a)216.238.XXX.XXX [BREAK] 2014-02-25 11:00:04.836881 [DEBUG] mod_sofia.c:127 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State Change CS_INIT -> CS_ROUTING 2014-02-25 11:00:04.836881 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/517NXXXXXX(a)216.238.XXX.XXX [BREAK] 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State INIT going to sleep 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) Running State Change CS_ROUTING 2014-02-25 11:00:04.836881 [DEBUG] sofia.c:5811 Channel sofia/internal/517NXXXXXX(a)216.238.XXX.XXX entering state [calling][0] 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State ROUTING 2014-02-25 11:00:04.836881 [DEBUG] mod_sofia.c:150 sofia/internal/517NXXXXXX(a)216.238.XXX.XXX SOFIA ROUTING 2014-02-25 11:00:04.836881 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-02-25 11:00:04.836881 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/517NXXXXXX(a)216.238.XXX.XXX [BREAK] 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State ROUTING going to sleep 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) Running State Change CS_CONSUME_MEDIA 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State CONSUME_MEDIA 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State CONSUME_MEDIA going to sleep 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.736724 [DEBUG] sofia.c:5811 Channel sofia/internal/888NXXXXXX(a)10.100.0.59 entering state [terminating][0] 2014-02-25 11:00:05.736724 [NOTICE] sofia.c:6655 Hangup sofia/internal/888NXXXXXX(a)10.100.0.59 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2014-02-25 11:00:05.736724 [DEBUG] switch_channel.c:3184 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [KILL]

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NetBSD Status
by Brian West 26 Feb '14

26 Feb '14

FreeSWITCHers, I’ve taken NetBSD as far as I can without some assistance from someone that is more familiar with NetBSD and various build issues that you may encounter. Here are some details if someone wishes to take this over the finish line: Configure detects the SWITCH_SIZE_T_FMT as “d”, It should be “zd” If you manually cane src/include/switch_am_config.h after configure it will compile. Here are a few of the patches that I’ve tweaked so far not sure if they are all that correct: http://www.bkw.org/netbsd.diff The presence of AC_FUNC_MALLOC in configure.in breaks malloc on NetBSD and rpl_malloc isn’t defined. NetBSD also suffers the same as a few other other BSD’s with a broken tgmath.h, thats the change to src/mod/applications/mod_spandsp/mod_spandsp_modem.h Also —disable-srtp will need to be used until the rest of the issues are resolved properly. Commit 378caebc9a1b67aa16ac69fc6b48452078ae19f1 solves the issue with disabling srtp. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

1 0

group call shared line
by Jason Holden 26 Feb '14

26 Feb '14

All, I have shared line working properly when dialing an extension directly, even if there are multiple registered extensions. The remaining problem is when group calling shared line does not functioning. Any recommendations to address this? Internal profile <param name="manage-presence" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="presence-hosts" value="$${domain},$${internal_sip_ip},$${external_sip_ip}"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/>

1 0

Brainstorming load balancing feature - dispatcher & FS
by jay binks 26 Feb '14

26 Feb '14

So I want to bash out a feature I was thinking of, please can anyone chime in some ideas on this. So its easy to have OpenSIP / Kamailio ( dispatcher ) send SIP Options to FS to see if its responding .. this is great ... My request would be to have some kind of control over the response FS Sends back. What I would love is to respond with idle CPU , current SPS / Max SPS or channels / Max Channels. This is only a high level "this would be nice" , so I havnt totally thought this through but... we would have to provide an ACL for who to provide this back to, coz I wouldnt really want any old guy on the internet sending me options and me respond going ... oh yea bro, Im at 90% ... dont have to try hard to DDOS Me.. :) but yea... if we could give this back to the load balancers, then we could easily instruct them to back traffic off to this gateway. the next part is , how do you back off the traffic. one option is to have FS respond differently if any of these are close to thresholds. or to provide these values in custom headers, but that would require changes in dispatcher ( which is ok I Guess ) anyways... I wanted some input on peoples thoughts here.. let me know what you think. Jay

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