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Stale channel that cannot be killed, session created from a running Lua scripts.
by Johny Kadarisman Kwan 11 Apr '13

11 Apr '13

I have lua 'background' script that wake up every few seconds, and check to perform specific task, one of it to established call. Calls are then initiated then transfer to some extension in dialplan. The scripts has been running and works as expected. But after session ended(hangup), channels is not released and can be seen using cli "show channels". and trying to kill such channel result in "-ERR No Such Channel!" This seems being reported before, but I can't seems to find any resolution. Following simple script could replicate the issues : freeswitch> luarun SimpleDialer.lua =================== SimpleDialer.lua local threadName = "SimpleDialer" while true do freeswitch.consoleLog("info", threadName.." ticking\n") -- task run here local new_session = freeswitch.Session('sofia/gateway/vox/+150<%2B15085895115> 88888888') new_session:transfer("echo", "XML", "default") freeswitch.msleep(60000) end ============= dialplan entry <extension name="echo"> <condition field="destination_number" expression="^echo$"> <action application="echo" data=""/> </condition> </extension>

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SRTP + ZRTP mixed modr
by Daniel Ivanov 11 Apr '13

11 Apr '13

Ok, i am getting confused. Am trying to offer mixed mode calls. If peers are compatible with zrtp(i check that in a db via mod_lua), then zrtp should be used. If it's not zrtp friendly or it's an outside call, then srtp is used. I set clients to create zrtp +optional srtp(e.g. both savp and avp in sdp lines) and then set via lua: Sip_allow_crypto_in_avp t Sip_secure_media f Nolocal:sip_secure_media f Zrtp_secure_media t Proxy_media t Bypass_media f Inherit_codec t But i only get the call established on clients with SRTP indicated as enabled and no sign of ZRTP. And there is no audio on top of that too, because fs is not feeding srtp data between them. Anyone has achieved such a dynamic mixed mode setup and wants to help?

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Detecting call loops while not relying on the hop counter
by Victor Chukalovskiy 11 Apr '13

11 Apr '13

Hello, I have a scenario where FS routes hundreds of calls and loop detection is needed. Adding to that, I can't rely on hop counter being decremented by other network elements. I can use limit in the following fashion: <action application="limit" data="hash loop_test ${caller_id_number}${destination_number} 10 !NORMAL_CIRCUIT_CONGESTION"/> So, whenever I have 10 calls with the same calling and called party number combination, limit will kick-in and break the loop. *Question*: Is this efficient from the perspective of how "limit" and hash back-end work? Once I have a few thousand calls, all having unique "resource" will this become resource hungry? My main concern is that it will create hundreds of hash entries all having the same realm but unique resource. Anyone knows a better solution? Thank you, Victor

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freeswitch on ipsec
by sibu 11 Apr '13

11 Apr '13

Dear Freeswitch-Users/Developers I am new to this list and freeswitch. I would like to know if anyone has tried freeswitch with an ipsec VPN such as openswan or strongswan and what were/should-be the settings (eg transport- mode, tunnel-mode etc), results and requirements vis a vis cpu-power, network-speed etc etc all hints and suggestions welcomed thanks in advance sibu xolo

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Outgoing fax issue.
by Grant Iler 11 Apr '13

11 Apr '13

Hi, We have a freeswitch setup with an Sangoma A200 plus 1 remora card setup. The cards have 6 FXO ports that go to 5 POTS lines and 2 FXS ports. Our fax machine is plugged into the first FXS port (freetdm 1:1). Incoming faxes work fine but outgoing faxes do not. The fax line is the 3rd Pots line (freetdm 2:3). The incoming faxes are bridged with this in the 00_inbound_did.xml: <!- LINE3 328-6982-> < !- Always directed to the fax machine -> < extension name=”Line 3(6982)” continue=”true”> < condition field=”source” expression=”mod_freetdm”/> < condition field=”${channel_name}” expression=”^(FreeTDM/2:3/)$”> < action application=”set” data=”domain_name=$${domain}”/> < action application=”set” data=”transfer_ringback=$${us-ring}”/> < action application=”bridge” data=”freetdm/2/3|freetdm/1/1″/> < /condition> < /extension> < !- LINE3 DONE-> Outgoing faxes are bridged with this in the dialplan/default.xml: <extension name=”bridge-FXS-to-FXO” continue=”true”> < condition field=”destination_number” expression=”^7(.+)$”/> < condition field=”channel_name” expression=”FreeTDM/1:1/”> < action application=”bridge” data=” freetdm/1/1/${destination_number}|freetdm/2/3/${destination_number}”/> </condition> < /extension> What I figured out from the logs and research on the wikis is that when faxes are sent out, the freeswitch server does not get the entire number through DTMF and the call fails. Here is a portion of the logs from the last outgoing fax that worked: 2013-04-04 11:18:23.494287 [DEBUG] ftmod_analog.c:830 [s1c1][1:7] Changed state from DIALTONE to COLLECT 2013-04-04 11:18:23.494287 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:23.494287 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7] 2013-04-04 11:18:23.514287 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from DIALTONE to COLLECT in 20 ms 2013-04-04 11:18:23.514287 [DEBUG] ftmod_analog.c:646 [s1c1][1:7] Executing state handler on 1:1 for COLLECT 2013-04-04 11:18:24.074292 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 1 (debug = 0) 2013-04-04 11:18:24.074292 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:24.074292 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71] 2013-04-04 11:18:25.113819 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 7 (debug = 0) 2013-04-04 11:18:25.113819 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:25.113819 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717] 2013-04-04 11:18:25.554309 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 5 (debug = 0) 2013-04-04 11:18:25.554309 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:25.554309 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7175] 2013-04-04 11:18:26.134309 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 7 (debug = 0) 2013-04-04 11:18:26.134309 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:26.134309 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71757] 2013-04-04 11:18:26.634314 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 2 (debug = 0) 2013-04-04 11:18:26.634314 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:26.634314 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717572] 2013-04-04 11:18:26.834315 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 2 (debug = 0) 2013-04-04 11:18:26.834315 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:26.834315 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7175722] 2013-04-04 11:18:27.614322 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 9 (debug = 0) 2013-04-04 11:18:27.614322 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:27.614322 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71757229] 2013-04-04 11:18:28.373722 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 0 (debug = 0) 2013-04-04 11:18:28.373722 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:28.373722 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717572290] 2013-04-04 11:18:28.894333 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 6 (debug = 0) 2013-04-04 11:18:28.894333 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:28.894333 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7175722906] 2013-04-04 11:18:29.714340 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 3 (debug = 0) 2013-04-04 11:18:29.714340 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:29.714340 [DEBUG] mod_freetdm.c:2488 got DTMF sig [71757229063] 2013-04-04 11:18:30.094343 [DEBUG] ftdm_io.c:3783 [s1c1][1:7] Queuing DTMF 4 (debug = 0) 2013-04-04 11:18:30.094343 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-04 11:18:30.094343 [DEBUG] mod_freetdm.c:2488 got DTMF sig [717572290634] 2013-04-04 11:18:32.114360 [DEBUG] ftmod_analog.c:851 [s1c1][1:7] Number obtained [717572290634] 2013-04-04 11:18:32.114360 [DEBUG] ftmod_analog.c:852 [s1c1][1:7] Changed state from COLLECT to RING 2013-04-04 11:18:32.134360 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from COLLECT to RING in 19 ms 2013-04-04 11:18:32.134360 [DEBUG] ftmod_analog.c:646 [s1c1][1:7] Executing state handler on 1:1 for RING This is what happens now: 2013-04-09 14:54:20.616457 [DEBUG] ftmod_analog.c:830 [s1c1][1:7] Changed state from DIALTONE to COLLECT 2013-04-09 14:54:20.616457 [DEBUG] mod_freetdm.c:2318 got FXS sig [COLLECTED_DIGIT] 2013-04-09 14:54:20.616457 [DEBUG] mod_freetdm.c:2488 got DTMF sig [7] 2013-04-09 14:54:20.635887 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from DIALTONE to COLLECT in 20 ms 2013-04-09 14:54:20.635887 [DEBUG] ftmod_analog.c:646 [s1c1][1:7] Executing state handler on 1:1 for COLLECT 2013-04-09 14:54:22.636477 [DEBUG] ftmod_analog.c:851 [s1c1][1:7] Number obtained [7] 2013-04-09 14:54:22.636477 [DEBUG] ftmod_analog.c:852 [s1c1][1:7] Changed state from COLLECT to RING 2013-04-09 14:54:22.656477 [DEBUG] ftmod_analog.c:640 [s1c1][1:7] Completed state change from COLLECT to RING in 20 ms Twice the Number obtained got more than one digit but it always dropped one at least one digit so that the fax number was not correct. I tried using start_dtmf and start_dtmf_generate from the Freeswitch wiki and also looked in the mailing list for issues with dtmf and faxing but was not able to find any solutions. I tried switching the FXS port to the 2nd one and swapped out the phone line but that did not have any other results. Let me know if you need to know more about the configuration or logs. Hopefully it is something easy someone with more knowledge than me can figure out. Thanks! Grant

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PRI channels suspended
by Ashish gautam 11 Apr '13

11 Apr '13

ftdm dump for channels 5 to 30 shows channels SUSPENDED. Here is the output: span_id: 1 chan_id: 5 physical_span_id: 1 physical_chan_id: 5 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: UP type: B state: SUSPENDED last_state: RESTART txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NORMAL_UNSPECIFIED session: (none) Please help me get out of this. Thanks. -Ashish

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Unloop extension
by Ержан Туле пов 11 Apr '13

11 Apr '13

What unloop extension is used for? BTW: vars.xml pre-(process)-sets unroll_loops, whereas DPs (public and default) are using channel variable syntax? Is this ok? Y.

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native support of postgres enable-core-pgsql-support and without-pgsql
by Yuriy Nasida 10 Apr '13

10 Apr '13

Hi guys, I had idea to try the new feature of native support of postgres today but it was not so easy as I though at the beginning. After installing of postgresql on my CentSO 6.2 (simple via yum) and trying to recompile FS with --enable-core-pgsql-support. I have got many different issues in the process of compiling. All them were well-known as far as I understand but it was still not easy to understand how to fix it and to get native postgres support. FS-3384, FS-3630, FS-3384 and FS-3393. I had all this problems.http://wiki.freeswitch.org/wiki/Installation_Guide#Release.28es.29… more full info. For solving was suggested ./configure --without-pgsql . But.... I would like to have pgsql in core... In the end I was tried this combination :)./configure --prefix=/usr/local/freeswitch_1_2_pg --enable-core-pgsql-support --without-pgsql So with core-pgsql and without pgsql simultaneously. It looks strange but it works. Hope it will help somebody else. I have one more question. Can I add postgres native support for registrations only ? Like I can have this by means of adding of dbc-dsn parameter to internal.xml only ? P.S. Very nice increase speed with native postgree . -- YN

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[ANN] CID Lookup and Fat Free CRM
by Sergey Zhuravlov 10 Apr '13

10 Apr '13

Hey Guys, We are excited to announce that Fat Free CRM is now available to be used with FreeSWITCH's mod_cidkookup. Solution based on standart mod_cidlookup and also may be used with Asterisk. FFCRM -- lightweight CRM used Ruby on Rails. http://www.fatfreecrm.com/ An open source CRM designed to be highly customizable; elegant in simplicity. Yet doesn't have any integration with PBX. Perfect system, especially for Rails' funs. Source code https://github.com/4orbit/ffcrmcidlookup and https://github.com/4orbit/fat_free_crm/tree/cidlookup See short article in Russian http://ru.it.ntcom.lv/blog/cid-lookup-ffcrm The future plans of closer integration with the FreeSWITCH: - wake up the customer's card for an incoming call - "smart" routing call to manager Supposed to use websocket! Supporters are welcomed -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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Do I need a new dns resource records for each tenant?
by jow 10 Apr '13

10 Apr '13

The subject says it all. do I need a new dns resource records for each new tenant? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Do-I-need-a-new-dns-resource-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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