March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ZRTP enrollment lua script question
by Peter Villeneuve 29 Mar '14

29 Mar '14

Hi all, I'm trying out the option to have FS play trusted MiTM for ZRTP. I just compiled Master from Git and tried calling the 9787 extension from my softphone with ZRTP. I see in the dialplan that the script appears to get executed, however I don't hear anything at all and after a little while the call hangs up. It's not a codec issue because calling the IVR works fine. So, what else do I need to do to allow ZRTP enrollment? I see in the wiki that I need to set <action application="set" data="zrtp_enrollment=true"/> So how do I set this globally so that FS always negotiates ZRTP with enabled clients? Or do I need to set that up in the dialplan? 98529894-b6c9-11e3-9ebf-efdbe42c5e9a Dialplan: sofia/internal/ 1000(a)my.domain.com Regex (PASS) [zrtp_enrollement] destination_number(9787) =~ /^9787$/ break=on-false 98529894-b6c9-11e3-9ebf-efdbe42c5e9a Dialplan: sofia/internal/ 1000(a)my.domain.com Action lua(zrtp_agent.lua) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:214 (sofia/internal/1000(a)my.domain.com) State Change CS_ROUTING -> CS_EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_session.c:1385 Send signal sofia/internal/1000(a)my.domain.com[BREAK] 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:523 (sofia/internal/1000(a)my.domain.com) State ROUTING going to sleep 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000(a)my.domain.com) Running State Change CS_EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:530 (sofia/internal/1000(a)my.domain.com) State EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] mod_sofia.c:178 sofia/internal/1000(a)my.domain.com SOFIA EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_core_state_machine.c:256 sofia/internal/1000(a)my.domain.com Standard EXECUTE 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000@my.domain.comhash(insert/my.domain.com-spymap/1000/98529894-b6c9-11e3-9ebf-efdbe42c5e9a) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000(a)my.domain.com hash(insert/my.domain.com-last_dial/1000/9787) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000@my.domain.comhash(insert/my.domain.com-last_dial/global/98529894-b6c9-11e3-9ebf-efdbe42c5e9a) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000(a)my.domain.com export(RFC2822_DATE=Fri, 28 Mar 2014 22:37:51 +0000) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.375885 [DEBUG] switch_channel.c:1245 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 28 Mar 2014 22:37:51 +0000] 98529894-b6c9-11e3-9ebf-efdbe42c5e9a EXECUTE sofia/internal/ 1000(a)my.domain.com lua(zrtp_agent.lua) 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.471868 [DEBUG] switch_core_media.c:343 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 d873ef12ce3143ba4ee41295c1b3199f49a89a78249ef454b3333a784d77acfd 98529894-b6c9-11e3-9ebf-efdbe42c5e9a 2014-03-28 22:37:51.471868 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [opus:111:48000:20:0]/[opus:116:48000:20:0] Thanks, Peter

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Freeswtich with freetdm - Span status is always "DOWN"
by Neo Cheema 29 Mar '14

29 Mar '14

I'm trying to use my ALLO Pri card with Libpri-Dahdi-FreeTDM stack. Here are the packages I've used Libpri - v1.4.14 Dahdi - v2.9.1 Freeswitch - Latest pull from Git All my efforts for Inbound/Outbound calls using PRI line are failing. freeswitch@caller0> originate FREETDM/1/1/998822xxxxx &park() 2014-03-29 14:59:44.346079 [INFO] ftmod_zt.c:671 Setting echo cancel to 64 taps for 1:1 2014-03-29 14:59:44.346079 [NOTICE] switch_channel.c:1053 New Channel FreeTDM/1:1/9988225250 [e6b2e0fd-fa62-43eb-8b4b-0a48c79b0c99] 2014-03-29 14:59:44.366092 [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! 2014-03-29 14:59:44.366092 [NOTICE] mod_freetdm.c:1772 Close Channel FreeTDM/1:1/9988225250 [CS_INIT] 2014-03-29 14:59:44.366092 [NOTICE] switch_ivr_originate.c:2707 Cannot create outgoing channel of type [FREETDM] cause: [DESTINATION_OUT_OF_ORDER] -ERR DESTINATION_OUT_OF_ORDER Somehow the channels are not getting configured properly as the status of Span1 is always DOWN. Any help would be greatly appreciated. I've tried the following commands for loading/debugging etc. freeswitch@caller0> load mod_freetdm 2014-03-29 14:51:06.806089 [INFO] mod_enum.c:876 ENUM Reloaded 2014-03-29 14:51:06.806089 [INFO] switch_time.c:1369 Timezone reloaded 530 definitions 2014-03-29 14:51:06.806089 [NOTICE] ftdm_sched.c:185 Launching main schedule thread 2014-03-29 14:51:06.806089 [NOTICE] ftdm_io.c:6344 Modules configured: 1 2014-03-29 14:51:06.806089 [NOTICE] ftmod_zt.c:1504 Using DAHDI control device 2014-03-29 14:51:06.806089 [INFO] ftdm_io.c:5516 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:601 Setting rxgain val to 0.000000 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:609 Setting txgain val to 0.000000 2014-03-29 14:51:06.806089 [INFO] ftdm_io.c:5480 Auto-loaded I/O module 'zt' 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:409 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:57 2014-03-29 14:51:06.806089 [INFO] ftmod_zt.c:409 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 1:2 fd:58 .......... 2014-03-29 14:51:06.826081 [INFO] ftdm_io.c:5419 Configured 124 channel(s) 2014-03-29 14:51:06.826081 [CONSOLE] switch_loadable_module.c:1464 Successfully Loaded [mod_freetdm] 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'freetdm' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'disable_ec' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'disable_dtmf' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:269 Adding Application 'enable_dtmf' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:315 Adding API Function 'ftdm' 2014-03-29 14:51:06.826081 [NOTICE] switch_loadable_module.c:315 Adding API Function 'ftdm_usage' +OK Reloading XML +OK freeswitch@caller0> ftdm list <empty> freeswitch@caller0> ftdm start 1 +OK freeswitch@caller0> ftdm dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: DOWN type: B state: DOWN last_state: DOWN txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) Dahdi command outputs root@caller0:/usr/local/freeswitch/bin# dahdi_scan [1] active=yes alarms=OK description=ALLO (PCI) Quad E1 Card 0 Span 1 name=Tor3/0/1 manufacturer= devicetype= location=PCI Bus 02 Slot 01 basechan=1 totchans=31 irq=0 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 root@caller:/usr/local/freeswitch# lsdahdi ### Span 1: Tor3/0/1 "ALLO (PCI) Quad E1 Card 0 Span 1" (MASTER) CCS/HDB3/CRC4 ClockSource 1 unknown Clear 2 unknown Clear 3 unknown Clear 4 unknown Clear 5 unknown Clear 6 unknown Clear 7 unknown Clear 8 unknown Clear 9 unknown Clear 10 unknown Clear 11 unknown Clear 12 unknown Clear 13 unknown Clear 14 unknown Clear 15 unknown Clear 16 unknown HDLCFCS 17 unknown Clear 18 unknown Clear 19 unknown Clear 20 unknown Clear 21 unknown Clear 22 unknown Clear 23 unknown Clear 24 unknown Clear 25 unknown Clear 26 unknown Clear 27 unknown Clear 28 unknown Clear 29 unknown Clear 30 unknown Clear 31 unknown Clear

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Playback case sensitive
by Grant Bagdasarian 28 Mar '14

28 Mar '14

Hello, Is there a way to make the playback application ignore case sensitivity of audio files it has to play? For instance: the actual file name is A.wav But when you do the following, it doesn't work because it can't find the file. <action application="playback" data="a.wav" /> It makes a sense that it doesn't. I'm just looking for a built-in freeswitch workaround/parameter/etc to get around this. Thanks, Grant

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Mod Limit
by Andrew 28 Mar '14

28 Mar '14

Hi, I am trying to set a limit to one channel on leg A. For some reason this is not causing the call to fail and disconnect. Anyone have any ideas? leg_A.Execute("limit", string.Format("hash outbound gw_{0} {1}!NORMAL_TEMPORARY_FAILURE", 1,1)); leg_A.Execute("limit", string.Format("hash outbound gw_{0} {1}/1!NORMAL_TEMPORARY_FAILURE", 1, 1)); leg_A.SetVariable("limit_ignore_transfer", "true"); <transfer_history>1396017461%3A39603950-fee8-4194-8420-7d259a0cd498%3Abl_xfe r%3Alimit_exceeded/public/XML</transfer_history> <transfer_source>1396017461%3A39603950-fee8-4194-8420-7d259a0cd498%3Abl_xfer %3Alimit_exceeded/public/XML</transfer_source> <limit_usage>2</limit_usage> <limit_usage_outbound_gw_1>2</limit_usage_outbound_gw_1> <limit_rate>1</limit_rate> <limit_rate_outbound_gw_1>1</limit_rate_outbound_gw_1> <limit_ignore_transfer>true</limit_ignore_transfer>

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Freeswitch/freetdm<->Freeswitch PCMU always wins
by Pete Ashdown 27 Mar '14

27 Mar '14

PSTN -> Freeswitch/freetdm (sending) -> Freeswitch/sip (receiving) PCMU is always winning between these two Freeswitch 1.2.23 boxes regardless of order in the codec list, when a call comes on freetdm. If I remove PCMU from the receiving box global_codec_prefs and thus the sofia external profile "CODECS IN" it works as expected and picks the top agreed codec from the freetdm sending box and the receiving. Before I paste debug logs, is there something I should be looking for on the sending box in relation to freetdm?

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UBUNTU VS WINDOWS
by Ryan Gholam 27 Mar '14

27 Mar '14

Hi , I have this problem that has been taking quite some time , I have 600 concurrent calls running on freeswitch in WINDOWS , and running on another pc (UBUNTU ) , i find that freeswitch on ubuntu is much more efficient than windows . The basic error that i am getting on Windows when running it for 10 hours is : [CRIT] mod_event_socket.c 2618 Socket Error! At Originate Result : *NORMAL_CLEARING, which is due to Socket Error .* *Is there some way i can have the same efficiency ? *

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Shared Presence and jira ticket FS-3758
by Chris B. Ware 27 Mar '14

27 Mar '14

Hi guys, I've exactly the same problem stated on jira ticket FS-3758. It seems that Anthony made a patch to fix it. Is it still there? How can activate that functionality? My knowledge of FS code is not enough to check by myself. Thanks in advance.

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Shared Presence and jira ticket FS-3758
by Chris B. Ware 27 Mar '14

27 Mar '14

Hi guys, I've exactly the same problem stated on jira ticket FS-3758. It seems that Anthony made a patch to fix it. Is it still there? How can activate that functionality? My knowledge of FS code is not enough to check by myself. Thanks in advance.

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ClueCon Tickets Are Live!
by Ken Rice 27 Mar '14

27 Mar '14

Hey FreeSWITCHers, ClueCon Ticket ordering is Live! Visit ClueCon.com and hit the registration link to get your tickets ordered today! ClueCon is Aug 4-7, 2014 at the Intercontinental Hotel in the heart of Chicago. What is ClueCon you ask? ClueCon is an an annual telephony and voice over IP developers conference started in 2005 by the FreeSWITCH core developer team. In fact FreeSWITCH itself was conceived at the first ClueCon as part of a large discussion between several open source leaders on how to improve the state of existing Internet Telephony options. Since then, every year ClueCon brings Telephony developers face to face with each other and with interesting new start-up and enthusiasts alike. It¹s a perfect blend of technical and business-oriented discussions from the leaders in the industry. The motto of the conference is ³A conference for developers, by developers² We focus on communication and getting people talking and sharing ideas. We have open discussions at the end of each presentation and a hearty lunch each day with plenty of time for meeting and speaking with the attendees and presenters. Each day is filled with interesting presentations in a relaxed atmosphere. You can watch a demo on the latest and greatest innovation in VoIP, or take a break and chat with your favorite open source developer or all of your friends that you usually only see online. ClueCon is the very best opportunity to get under the hood of many telephone related products and learning from the actual developers who made them possible. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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What is the way to send gateway name from mod_gsmopen to softphone?
by minas_trith 26 Mar '14

26 Mar '14

Sorry for my english. I use the FreeSWITCH 1.5.8b+git~20140202T113337Z~9059fb91cc~64bit (Wheezy 64bit) and mod_gsmopen (13xE1550) for a month. Now I must identify gateway/line which is calling for extension. The name gateway (gsmopen: gsm01, gsm02 ...) must be displayed on softphone (xlite, jitsi) together with the number of callers. What is the way/mod etc... that will make it possible? Przemysław Pereć

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