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Correct way to hangup a channel
by Stephen Wilde 11 Nov '11

11 Nov '11

I have the necessity to hangup a channel inside a media bug callback in a my custom module. Currently, what I do is: static switch_bool_t my_media_callback(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type) { struct my_custom_helper * my_helper = (struct my_custom_helper *) user_data; switch (type) { case SWITCH_ABC_TYPE_READ_REPLACE: case SWITCH_ABC_TYPE_WRITE_REPLACE: { switch_frame_t *frame; if (sth->read) { frame = switch_core_media_bug_get_read_replace_frame(bug); } else { frame = switch_core_media_bug_get_write_replace_frame(bug); } /* .... media analysis .... */ if (to_hangup) { switch_channel_t *channel = switch_core_session_get_channel(my_helper->session); if (channel) { switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING); } } } return SWITCH_TRUE; } The dropped channel is a b-leg, previously bridged with an a-leg (incoming call). Probably in my code there is an error because the code works fine and the channel is correctly dropped but inside Freeswitch remain some "zombies" calls. If I issue a fs_cli "show channels" I see that there are some calls dropped by my module showed in "ACTIVE" state. What is wrong with my channel hangup? Thanks in advance Stephen

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freeTDM can't configure with Digium TDM410 card
by Melvin King 11 Nov '11

11 Nov '11

ok, here's my setup. TDM410 card with port 1 having a line to the PSTN and ports 2-4 have the telephone lines plugged in FS version is 1.0.head (git-9dd45e3 2011-11-03 14-45-01 -0500) freetdm.conf has full rw permissions My problem is that I cannot configure freetdm to use this blasted card, no matter what I do..... Assistance appreciated this is my freetdm.conf file [span zt FXS] trunk_type => FXS fxs-channel => 1 [span zt FXO] trunk_type => FXO fxo-channel => 2-4 this is the relevant output from the logfile 2011-11-10 14:16:02.229171 [DEBUG] ftdm_config.c:52 New mod directory: /usr/local/freeswitch/mod 2011-11-10 14:16:02.229411 [DEBUG] ftdm_config.c:58 New config directory: /usr/local/freeswitch/conf 2011-11-10 14:16:02.229656 [DEBUG] ftdm_sched.c:154 Initializing scheduling API 2011-11-10 14:16:02.229732 [DEBUG] ftdm_sched.c:251 Created schedule freetdm-master 2011-11-10 14:16:02.229792 [NOTICE] ftdm_sched.c:178 Launching main schedule thread 2011-11-10 14:16:02.229938 [DEBUG] ftdm_sched.c:187 Running schedule freetdm-master in the main schedule thread 2011-11-10 14:16:02.230029 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/modules.conf. 2011-11-10 14:16:02.230288 [NOTICE] ftdm_io.c:5725 Modules configured: 1 2011-11-10 14:16:02.230494 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/freetdm.conf. 2011-11-10 14:16:02.230592 [DEBUG] ftdm_io.c:4635 Reading FreeTDM configuration file 2011-11-10 14:16:02.230687 [DEBUG] ftdm_io.c:4651 found config for span 2011-11-10 14:16:02.231224 [NOTICE] ftmod_zt.c:1323 Using DAHDI control device 2011-11-10 14:16:02.231410 [INFO] ftdm_io.c:4965 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt] 2011-11-10 14:16:02.231589 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/zt.conf. 2011-11-10 14:16:02.231826 [INFO] ftmod_zt.c:585 Setting rxgain val to 0.000000 2011-11-10 14:16:02.231922 [INFO] ftmod_zt.c:593 Setting txgain val to 0.000000 2011-11-10 14:16:02.232044 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt' 2011-11-10 14:16:02.232201 [DEBUG] ftdm_io.c:4672 created span 1 (FXS) of type zt 2011-11-10 14:16:02.232274 [DEBUG] ftdm_io.c:4690 span 1 [trunk_type]=[FXS] 2011-11-10 14:16:02.232370 [DEBUG] ftdm_io.c:4695 setting trunk type to 'FXS' 2011-11-10 14:16:02.232517 [DEBUG] ftdm_io.c:4690 span 1 [fxs-channel]=[1] 2011-11-10 14:16:02.232770 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 1 fd 16 (Invalid argument)] 2011-11-10 14:16:02.233387 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 1 as FreeTDM device 1:1 fd:17 2011-11-10 14:16:02.233595 [DEBUG] ftdm_io.c:5263 Creating new group:__default 2011-11-10 14:16:02.233746 [DEBUG] ftdm_io.c:4651 found config for span 2011-11-10 14:16:02.233931 [DEBUG] ftdm_io.c:4672 created span 2 (FXO) of type zt 2011-11-10 14:16:02.234045 [DEBUG] ftdm_io.c:4690 span 2 [trunk_type]=[FXO] 2011-11-10 14:16:02.234111 [DEBUG] ftdm_io.c:4695 setting trunk type to 'FXO' 2011-11-10 14:16:02.234173 [DEBUG] ftdm_io.c:4690 span 2 [fxo-channel]=[2-4] 2011-11-10 14:16:02.234550 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 2 fd 16 (Invalid argument)] 2011-11-10 14:16:02.235149 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 2 as FreeTDM device 2:1 fd:18 2011-11-10 14:16:02.235550 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 3 fd 16 (Invalid argument)] 2011-11-10 14:16:02.236157 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 3 as FreeTDM device 2:2 fd:19 2011-11-10 14:16:02.236544 [WARNING] ftmod_zt.c:352 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 4 fd 16 (Invalid argument)] 2011-11-10 14:16:02.237129 [INFO] ftmod_zt.c:404 configuring device /dev/dahdi/channel channel 4 as FreeTDM device 2:3 fd:20 2011-11-10 14:16:02.237385 [INFO] ftdm_io.c:4887 Configured 4 channel(s) 2011-11-10 14:16:02.237644 [ERR] mod_freetdm.c:2953 open of freetdm.conf failed 2011-11-10 14:16:02.238048 [DEBUG] ftdm_sched.c:217 Waiting for main schedule thread to finish 2011-11-10 14:16:02.333760 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ... 2011-11-10 14:16:02.338252 [INFO] ftdm_io.c:601 Closing channel zt:1:1 fd:17 2011-11-10 14:16:02.338393 [INFO] ftdm_io.c:601 Closing channel zt:2:1 fd:18 2011-11-10 14:16:02.338528 [INFO] ftdm_io.c:601 Closing channel zt:2:2 fd:19 2011-11-10 14:16:02.338601 [INFO] ftdm_io.c:601 Closing channel zt:2:3 fd:20 2011-11-10 14:16:02.338720 [INFO] ftdm_io.c:5092 Unloading I/O interface zt 2011-11-10 14:16:02.338787 [INFO] ftdm_io.c:5099 Unloaded I/O interface zt 2011-11-10 14:16:02.338848 [INFO] ftdm_io.c:5118 Unloading module /usr/local/freeswitch/mod/ftmod_zt.so 2011-11-10 14:16:02.338997 [DEBUG] ftdm_dso.c:90 lib 0xa103d78 was closed with success 2011-11-10 14:16:02.339062 [INFO] ftdm_io.c:5120 Unloaded module /usr/local/freeswitch/mod/ftmod_zt.so 2011-11-10 14:16:02.339124 [DEBUG] ftdm_sched.c:552 Destroying schedule freetdm-master 2011-11-10 14:16:02.339230 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_freetdm.so **Module load routine returned an error**

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paramter "NDLB-force-rport" did not effect
by fieldpeak 11 Nov '11

11 Nov '11

Dear Friends, i met a problem that call a registered user behind NAT, i need FS send INVITE message to the port of the NAT device from which the registered user sent out REGISTRATION but not according to the CONTACT header. see the wiki, i enable below paramter, <param name="NDLB-force-rport" value="true"/> >From the wireshark trace, it shows FS received the REGISTERATION message from remote port 20438, however, the INVITE message still sent to the port(20347) inside "Contact", it looks the paramter "NDLB-force-rport" did not effect, Could anyone help for this issue, thanks a lot! Below is the registeration info from FS output (press F9), Call-ID: 1721091027(a)10.33.48.203 User: 13580358068(a)124.193.106.104 Contact: "user" <sip:13580358068@61.144.248.77:20347> Agent: Status: Registered(UDP)(unknown) EXP(2011-11-08 14:38:02) EXPSECS(152) Host: freeswitch IP: 61.144.248.77 Port: 20438 Auth-User: 13580358068 Auth-Realm: 124.193.106.104 MWI-Account: 13580358068(a)124.193.106.104 Cheers! BR, Charles

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Playlist playback
by Hynek Cihlar 11 Nov '11

11 Nov '11

Does Freeswitch support playing back of multiple audio files? I.e. issuing one command with multiple files? I am trying to build dynamic dialogs over event socket and issuing call for every part of the dialog seems as a waste. Sent from my mobile device

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Call Broadcasts - when to start playing?
by Avi Marcus 11 Nov '11

11 Nov '11

I've got a simple message that a business needs to play for all his contacts. How do you folks do broadcasts? There's the commerical mod_amd but I don't think I'm getting that just yet. wait_for_silence for 2 seconds (to wait out the answering machine?), then play the message? How many "intervals" is 2 seconds? Or mix wait_for_silence with mod_avmd via ESL so either one triggers it? Or does mod_avmd do both the beep (voicemail still beeps??) and the silence? Thanks for your input, -Avi

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I missing conference moderator features
by Thomas Hoellriegel 10 Nov '11

10 Nov '11

Hi all, I have a few suggestions as to improve the usability conferences as moderator. For example: you press a key, and become these menuoptions: with 1 and 3, they can search through the channels in lissen only mode. press 2 for the current channel. press 4 to bridge the current channel with your self. press 5 to mute the current channel. press 6 to kick the channel from the conference. Press 7 to move the current channel to another conference-number. press the 8 to the subscriber number of the user to hear. So you can simply remove disturb as moderator of the conference soon. eavdrop has only limiting functions. Unfortunately, I find no application where this enables channel-search mode. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Sofia Recover not retaining record route
by Gilad Abada 10 Nov '11

10 Nov '11

Hi all I am using sofia recover with unregistered trunks and on recovery Freeswitch seems to lose the record route and VIA headers, before the recover FreeSwitch has all the info. After the recover I get the call back but FreeSwitch is still trying to recover and then ultimately disconnects the call with this error: 2011-11-10 13:48:23.227240 [NOTICE] sofia.c:6039 Hangup sofia/external/+1XXXXXXXXXX(a)CARRIER.com [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] The reINVITE to "salvage" the call is missing the record-route info. I have a capture that I can send for further trouble shooting. This test was done on an incoming call. Thanks so much, Gill -- Gilad Abada SteadFast Telecommunications, Inc. V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office.

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reporting bug for freeswitch-contrib project
by Hector Geraldino 10 Nov '11

10 Nov '11

Hi, What's the best way to report a bug (and a patch) for a project that is part of the freeswitch-contrib repository? I can't find any category on JIRA for those projects. Should I send the bug description and the proposed patch to the freeswitch-dev list, or is there a better way? Thanks

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Help with home setup
by Jacob Smith 10 Nov '11

10 Nov '11

I have been trying to get this working for the last week and would really appreciate your help. I have a pfSense 2.0 server with Freeswitch and FusionPBX installed. I followed the instructions at http://wiki.fusionpbx.com/index.php?title=PfSense_Install and can access the web GUI. My goal is to use a Gigaset C610A-IP connected to my Google Voice account. However, my first failure seems to be the most basic: through FusionPBX, I created a user and extension to use with the phone but I can not register it with FreeSwitch. So, how do I get the phone to work so I can test that the basic functions of FreeSwitch are working and move on? Thanks!

2 2

Voice mail migration
by Lloyd Aloysius 10 Nov '11

10 Nov '11

HI, I am in the process of moving my current freeswitch server to a new server. What is the best way to migrate the VoiceMail from old to new server? Any help is appreciated. Thanks and regards, Lloyd

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