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mod_fifo WITHOUT answering the call during queue
by Michael Huber 20 Jul '14

20 Jul '14

Hey there! I was wondering if there is any possibilty to use "mod_fifo" but do NOT answer a call immediately. Instead there should be just a ringing to the originator while being in the queue but NO answer to the call until an agent pick up the call. Can anybody help? Best Michael

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LUA pattern match question
by Sean Devoy 19 Jul '14

19 Jul '14

Hi All, I have modified this excellent example LUA script (https://confluence.freeswitch.org/display/FREESWITCH/Lua+Intercom+example) to return "true" if an extension is in use and "false" otherwise". Just FYI: this is great for Cisco phones because an auto-answer (intercom) connection will interrupt the current call unless you block ALL intercom use. With this I can sent intercom if not in use, or send as a standard call otherwise. My complements to the original chef! My request is for someone to explain the pattern matching pattern used in this function: function trim (s) return (string.gsub(s, "^%s*(.-)%s*$", "%1")) end It takes a string like this: "uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num d7bd917c-efca-4333-81f6-692c727069a2,inbound,2014-07-18 11:59:41,1405699181,sofia/external_noauth/4107673000@66.241.96.237,CS_EXECUTE,+14107673000,4107673000,66.241.96.237,BOOKKEEPING,voicemail,default voicemail_ESTA 40,XML,from-internal-ESTA,L16,8000,128000,PCMU,8000,64000,,fs_bfis.bizfocused.com,4107673000(a)66.241.96.237,,ACTIVE,,,,,, 1 total." or just this: " 0 total." And returns just the number and the "Total." at the end, like "1 total." Or "0 total." I am fairly regex pattern savvy so I understand pattern matching, but I cannot decipher this LUA pattern. I see it works, but I would like to understand how/why. Thanks as always, Sean

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Skypopen.ko on VPS
by farzad jahangiri sisakht 19 Jul '14

19 Jul '14

Hello everyone. I'm wondering if anyone has managed to install the skypopen.ko sound driver on a VPS? I even tried with this error: rm -rf *.o *~ core .depend .*.cmd *.ko *.mod.c .tmp_versions make -C /lib/modules/2.6.32-042stab084.26/build M=/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss LDDINC=/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/../include modules make: *** /lib/modules/2.6.32-042stab084.26/build: Not a directory. Stop. make: *** [modules] Error 2 and even followed the link on the wiki : http://failverse.com/compiling-kernel-modules-for-a-rackspace-cloud-server/ but when I issue: zcat /proc/config.gz | sudo tee /usr/src/linux/.config > /dev/null I get: gzip: /proc/config.gz: No such file or directory tee: /usr/src/linux/.config: Not a directory Any help would be much appreciated.

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FS crash on call with mod_opal
by Zonzon Smith 19 Jul '14

19 Jul '14

Good day freeswitchers, I'm setting up a trunk to an Avaya PBX. I've successfully build mod_opal unfortunately calls either inbound or outbound make FS crash.I'm running FreeSWITCH (Version 1.4.7 git 59bdbec 2014-07-15 15:33:52Z 64bit) . Thank you. S.Z

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api - bgapi from ESL are redundant and confuse
by Normando Hall 19 Jul '14

19 Jul '14

Hi guys After 2 days I understand some issues with ESL and the NodeJS ESL wrapper (modesl). Learning to use event socket I read that has commands like: api - bgapi - linger, nolinger etc. Then, when reading the ESL wrapper for NodeJS I can't found those sockets commands. Only found api and bgapi, but not linger, nolinger, etc. So, the wrapper has the same methods as ESL has. But why I can't found those linger and nolinger methods? Here is the answer from the ESL wiki: *api($command, $args) is identical to sendRecv("api $command $args")* and *bgapi($command, $args) is identical to sendRecv("bgapi $command $args")* So... I can figured that "linger" or other commands should send into the "send" or "sendRecv" methods. This cause me a confussion. So, all ESL commands must be send through "send" or "sendRecv" methods, and "api" or "bgapi" are shorthands of "send" method for api/bgapi. I think these methods (api-bgapi) should be remove from ESL, to avoid future confussion. Regards Normando *//*

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Issue with make current - Error Connection refused
by Mike Burlingame 19 Jul '14

19 Jul '14

Anyone else having this issue? Pulling updates... fatal: unable to connect to git.freeswitch.org: git.freeswitch.org[0: 198.22.64.251]: errno=Connection refused

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Git Master missing header files
by Andrew 19 Jul '14

19 Jul '14

Hi, Today I downloaded master using git clone https://stash.freeswitch.org/scm/fs/freeswitch.git And lot of the headers are missing. Anyone know what I am doing wrong? I am using github on windows. Andrew

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Is it possible to force FreeSWITCH/Sofia to use only one port TLS?
by Trever L. Adams 19 Jul '14

19 Jul '14

https://wiki.freeswitch.org/wiki/SIP_TLS#Limitations_of_the_Freeswitch_TLS.… mentions that lib_sofia and FreeSWITCH uses two ports, by default, for TLS (client -> server for registration and client to server signaling, and server->client for NOTIFYs, etc.). In one setup, I need to get CSIPSIMPLE (an Android SIP client with ZRTP support) working. It appears to accept the server->client connections, but will not ring, etc. for incoming calls. This does seem to be a bug there, and it only exists with TLS (not tcp/udp). So, is it possible to force FreeSWITCH/Sofia to use the client->server connection for server->client NOTIFYs? Thank you, Trever

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Mod_callcenter ring in queue call agent
by OverVoIP LatAm 18 Jul '14

18 Jul '14

Hi, Just wondering if someone has tried it, we're trying to mod_callcenter to ring all agents even if they are already on a queue call. We already tried to do it adding more contact information like <agent name="agent1@default" type="callback" contact="[origination_caller_id_name='Queue Call',call_timeout=60,bridge_generate_comfort_noise=true]user/agent1@fsip,[origination_caller_id_name='Queue Call',call_timeout=60,bridge_generate_comfort_noise=true]sofia/internal/agent1%fsip,[origination_caller_id_name='Queue Call',call_timeout=60,bridge_generate_comfort_noise=true]user/agent1" status="Available" max-no-answer="1000" wrap-up-time="1" reject-delay-time="3" busy-delay-time="2" no-answer-delay-time="3"/> but if they're in queue call new calls just don't ring, We're looking something like ringinuse=yes in asterisk or something similar. If someone has done something similar we'll appreciate your help Regards overvoiplatam

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[OT] LVUG, the London VoIP User Group
by Richard Brady 18 Jul '14

18 Jul '14

Hi folks LVUG is a monthly get-together in London to discuss open-source VoIP technology, including FreeSWITCH. If you're based in or near London please join us this coming Tuesday for the third LVUG which will be held in Old Street. The format is a couple short talks followed by beers at the venue or in a nearby pub. Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/ Hope to see you there! Richard -- Richard Brady E: rnbrady(a)gmail.com

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