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ftdm console commands not responding as expected
by bedgar＠vseinc.com 22 Apr '13

22 Apr '13

Hi again freetdm Experts! There are many options for ftdm in the FS console: ftdm list ftdm start <span_id|span_name> ftdm stop <span_id|span_name> ftdm reset <span_id|span_name> [<chan_id>] ftdm alarms <span_id> <chan_id> ftdm dump <span_id|span_name> [<chan_id>] ftdm sigstatus get|set <span_id|span_name> [<chan_id>] [<sigstatus>] ftdm trace <path> <span_id|span_name> [<chan_id>] ftdm notrace <span_id|span_name> [<chan_id>] ftdm gains <rxgain> <txgain> <span_id|span_name> [<chan_id>] ftdm dtmf on|off <span_id|span_name> [<chan_id>] ftdm queuesize <rxsize> <txsize> <span_id|span_name> [<chan_id>] ftdm iostats enable|disable|flush|print <span_id|span_name> <chan_id> ftdm ioread <span_id|span_name> <chan_id> [num_times] [interval] ftdm core state [!]<state_name> ftdm core flag [!]<flag-int-value|flag-name> [<span_id|span_name>] [<chan_id>] ftdm core spanflag [!]<flag-int-value|flag-name> [<span_id|span_name>] ftdm core calls If I do a ftdm list I get: freeswitch@internal> ftdm list +OK span: 1 (wp1) type: analog_em physical_status: ok signaling_status: UP chan_count: 24 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none I would like to be able to stop this span but get errors trying to do it. freeswitch@internal> ftdm stop wp1 -ERR failure I think I am using the right name because I can restart the span and restart a single channel in the span. freeswitch@internal> ftdm reset wp1 Resetting channel wp1:1 Resetting channel wp1:2 Resetting channel wp1:3 Resetting channel wp1:4 Resetting channel wp1:5 Resetting channel wp1:6 Resetting channel wp1:7 Resetting channel wp1:8 Resetting channel wp1:9 Resetting channel wp1:10 Resetting channel wp1:11 Resetting channel wp1:12 Resetting channel wp1:13 Resetting channel wp1:14 Resetting channel wp1:15 Resetting channel wp1:16 Resetting channel wp1:17 Resetting channel wp1:18 Resetting channel wp1:19 Resetting channel wp1:20 Resetting channel wp1:21 Resetting channel wp1:22 Resetting channel wp1:23 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c1][1:1] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c2][1:2] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c3][1:3] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c4][1:4] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c5][1:5] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c6][1:6] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c7][1:7] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c8][1:8] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c9][1:9] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c10][1:10] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c11][1:11] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c12][1:12] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c13][1:13] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c14][1:14] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c15][1:15] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c16][1:16] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c17][1:17] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c18][1:18] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c19][1:19] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c20][1:20] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c21][1:21] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c22][1:22] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c23][1:23] VETO state change from DOWN to RESET freeswitch@internal> ftdm reset wp1 1 Resetting channel wp1:1 2013-04-19 19:33:24.838965 [WARNING] mod_freetdm.c:4656 [s1c1][1:1] VETO state change from DOWN to RESET Thanks for looking and helping if you can! Brian C. Edgar, Jr.

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session_locate and deadlocks
by Michel Brabants 21 Apr '13

21 Apr '13

Hello, I'm trying to passthrough some events from one leg of a bridged session to the other leg: notify-talk and hold. talk was an easy one. However, regarding the hold, I'm afraid of deadlocks, because I will be modying the peer-session (putting it on hold), which will also try to modify the originating leg (play hold-music). So, can a session get a readlock on a session that is currenly processing a request? I have the impression that there are higher-level-locks on sip-level. So, my question is: What is the function of the readlock created by "switch_core_session_locate" ? And, can multiple locks be acquired (think so), but what should I watch out for then? Thanks, Michel

2 2

Security considerations for configuration of a live internet system
by Topo Lino 21 Apr '13

21 Apr '13

Alright, I need use external tools as fail2ban and better use TLS, but in configuration... I understand the security configuration is on "sip_profiles". My current configuration is: <param name="auth-calls" value="true"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="log-auth-failures" value="true"/> Is there some other option recommended to use on "sip_profiles"? On the directory. Is it possible use only "user id" for authentication and don't accepts "number-alias" as username? Thanks. -- Localnet <http://www.localnet.org.es/>

1 0

FreeSwitch 1.3.7 Final - mod_freetdm Compilation Failure
by Arpit Gadle 21 Apr '13

21 Apr '13

Hi All, I had downloaded *freeswitch-1.3.17-final.tar.gz *from http://git.freeswitch.org/git/freeswitch/. ./bootstrap.sh and ./configure steps were completed successfully but make command failed with following error message *CC ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:55,* * from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:77: error: ‘SCT_MAX_NET_ADDRS’ undeclared here (not in a function)* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:55,* * from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:105: error: ‘MW_MAX_NUM_OF_PEER’ undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:109: error: ‘MW_MAX_NUM_OF_INTF’ undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:112: error: ‘MW_MAX_NUM_OF_CLUSTER’ undeclared here (not in a function)* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:129: error: expected declaration specifiers or ‘...’ before ‘SbMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:130: error: expected declaration specifiers or ‘...’ before ‘MwMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_m2ua.h:131: error: expected declaration specifiers or ‘...’ before ‘NwMgmt’* *In file included from src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:35:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:804: error: expected declaration specifiers or ‘...’ before ‘MwMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:805: error: expected declaration specifiers or ‘...’ before ‘NwMgmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:806: error: expected declaration specifiers or ‘...’ before ‘HiMngmt’* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_main.h:807: error: expected declaration specifiers or ‘...’ before ‘SbMgmt’* *cc1: warnings being treated as errors* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c: In function ‘sngss7_save_iam’:* *src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_support.c:3340: error: format ‘%d’ expects type ‘int’, but argument 10 has type ‘size_t’* *make[6]: *** [ftmod_sangoma_ss7_la-ftmod_sangoma_ss7_support.lo] Error 1* *make[5]: *** [../libfreetdm.la] Error 2* *make[4]: *** [all] Error 1* *make[3]: *** [mod_freetdm-all] Error 1* *make[2]: *** [all-recursive] Error 1* *make[1]: *** [all-recursive] Error 1* *make: *** [all] Error 2* * * Any help is appreciated. Thanks & Regards, Arpit Gadle

2 1

ftmod_zt.c error
by Ashish gautam 21 Apr '13

21 Apr '13

Hi, I am getting this error on the console. What could be the possible reasons for this and how to get rid of that: 2013-04-11 11:08:53.294506 [WARNING] ftmod_zt.c:1350 [s1c16][1:16] Event 6 is not dmtf related. Skipping one media read cycle 2013-04-11 11:08:53.294506 [WARNING] ftmod_zt.c:1352 [s1c16][1:16] Skipping one IO read cycle due to events pending in the driver queue 2013-04-11 11:08:53.294506 [ERR] ftmod_zt.c:1198 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2013-04-11 11:08:53.294506 [ERR] ftmod_zt.c:1198 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) Thanks in advance. -- Ashish

2 1

Respect the request URI domain and To URI domain when bridging
by José Luis Millán 21 Apr '13

21 Apr '13

Hi all, Every call bridged though a specific 'gateway' does rewrite the SIP request URI domain and To URI domain values to the gateway's 'proxy' parameter value. I would like to respect such values and receive in my proxy the SIP messages with their original request URI domain and To URI domain. This is, I would like to be able to send a SIP message with the following request URI and To headers, to the IP address 2.2.2.2 ``` INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:d@1.2.3.4>``` ``` Is it possible to handle this with FreeSwitch? I have followed the FS wiki but I've been unable to handle this. Thanks in advance. -- José Luis Millán

4 10

Mod_xml_radius features
by Dmitry Erankevich 20 Apr '13

20 Apr '13

Dear friends, I’m researching the freeswitch’s deploying perspective in our current infrastructure. One of the case is integration with a billing system. I have tried various modules but cannot find dynamic registration feature. And maybe Mod_xml_radius is what I need but unfortunately cannot understand how it works. Could you please help me to find out if: 1. Mod_xml_radius provides sip dynamic registration with Radius protocol. No xml files need. User database can be located in Radius server side. 2. It is possible to mix somehow radius registration with regular xml files. 3. With Mod_xml_radius is possible to apply maximum concurrent calls for the user. Thank you very much for your support, Dmitry.

1 0

Voicemail ends record after 30 seconds
by Pascal Taube 20 Apr '13

20 Apr '13

7 11

Audio Lost after 1 min
by Sean Devoy 20 Apr '13

20 Apr '13

Hi All, I have a report by a customer at a site that we had considerable trouble with NAT issues. I thought everything was now stable, but it sounds like it is not. She reports that she is only telling me now, but for some time when they call someone and get no answer and get directed to voicemail (the latest was a Sprint number) the audio from the remote answering machine cuts off before then end of the message. She did point out the call timer on her phone continued normally. She swears this does not happen when talking to a live person! First, I don't think FS can even tell if the b-leg was answered by voice mail differently from a person. Please correct me if that is incorrect. So, I am proceeding on the assumption that calls that are answered take less than 60 seconds! 45 seconds of the reported call were waiting for Sprint to give up on the cell phone and route to voicemail. The log just shows the call being answered . then Normal Clearing. I tend to always suspect NAT or crappy Polycom 335s for this customer as that has been all my issues with them so far. Is there a NAT issues relating to max call length, etc? No other users (all cisco phones) on the same FS at other sites report this issue. Thanks in advance. Sean

3 3

reloading mod_directory profiles
by Ira Tessler 20 Apr '13

20 Apr '13

Is there a way to reload the directory.conf (directory profiles) without restarting the server or doing a reload mod_directory? I use xml_curl to delivery the directory profiles and I need to by able to add new profiles on the fly. Thanks. Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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