July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Freeswitch shared registration
by Nick Zaitsev 18 Jul '14

18 Jul '14

Good day to you! Could you advise me with Freeswitch, please? I have load balancer (opensips) in front of 2 Freeswitch (fsw1 and fsw2) servers, each with their own hostname. The load balancing works fine, but the shared database did not work. I use PostgreSQL I add to external and internal sip profiles and sofia.conf.xml this: <param name="track-calls" value="true"/> <param name="core-db-dsn" value="pgsql://hostaddr=x.x.x.x dbname=db_name user=user password='password'"/> And to switch.conf.xml at fsw1: <param name="switchname" value="fsw1"/> <param name="track-calls" value="true"/> <param name="core-db-dsn" value="pgsql://hostaddr=x.x.x.x dbname=db_name user=user password='password'"/> <param name="core-recovery-db-dsn" value="pgsql://hostaddr=x.x.x.x dbname=db_name user=user password='password'"/> and at fsw2 the same,except hostname. There is not any errors, but the DB is empty. What I do wrong? Could you help me please? Thank you for your time, Best Regards.

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Clicks on line
by Paul Collister 18 Jul '14

18 Jul '14

Hi everyone, I wonder if anyone can point me in the right direction for fixing this problem. Im getting random clicks/glitches when I call the mw tone via a PSTN Gateway. This only started when I moved my installation to a high end server in a main hosting site. So this has excellent connectivity to the PSTN gateway. The server is a dedicated Dell Quad core 3ghz powerdge R210, with Centos 6.5 x.64. I don’t have any problem when I call my test server , which is a lower spec , on an adsl line, from the same gateway, Freeswitch runs perfectly and the tone is 100% pure. I have stopped all processes on the server I don’t need, but this had no effect, I have tried reducing the ethernet buffers with no effect. I suspect the server is sending out Mark bits at the wrong time and this is upsetting the gateway. Any pointers as to where to look would be much appreciated. All FS software is up to date Paul Collister

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 18 Jul '14

18 Jul '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday BugHunt Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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How to get notified when a conference recording ends?
by bernhard＠zeppelindesk.com 18 Jul '14

18 Jul '14

Hi all, what I want to archieve is recording a conference call in MP3 Format. As FS only allows for WAV recordings, I'm happy to transcode to MP3 after the recording has stopped. Recording will be started via ESL using "conference conf_name recording start /recording/filename.wav". Now there are 2 possible ways of how to end recording. Stopping recording via ESL (the easy one as the control application can start transcoding immediately afterwards). The conference ends as all participants leave (the hard one) The "record_post_process_exec_app" variable (*1) doesn't seem to work. The set application is never executed, neither on 1 nor on 2. Is there any other way to be notified about a finished conference recording other than to check the participants count for each recorded conference on a regular basis? Regards, Bernhard *1 [ https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Post_P… ]( https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Post_P… )

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detect dtmf on b leg during bridge
by Grant Bagdasarian 18 Jul '14

18 Jul '14

Hello, I'm using the group_confirm application to have the destinations accept the call before it's being connected to the a leg. So far, this is easy to implement. I have the following requirement, but I have no idea how to approach this in FS. 1) When the destination answers, it has to press a 1 to accept the call or 9 to bridge to another destination. So basically, a 1 or 9 should trigger the other ringing or answered destinations to be hungup. 2) When the call has been accepted, the b leg has the option to press 9 anytime during the conversation to bridge to another destination. I'm having a hard time thinking of a solution to implement this using the standard dialplan XML. Do I need to use ESL considering the complex nature of this or can it be done with existing XML tools? Regards, Grant

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Any one have experience for Asterisk register on freeswitch ?
by Shaw Xu 17 Jul '14

17 Jul '14

Thank you *Varghese Paul* for looking my post. My topology is like this: SIP trunk provider ---------- my FreeSwitch -------------- Elastix ------------ X-Lite Today, I make it work on the public IP asterisk registered to public IP FreeSwitch. original issue might be I put the wrong register string in Elastix. ( should be username:passwd@realm and yesterday was username:passwd@realm/username ) Means I can accept the phone from SIP trunk provider to FreeSwitch, and forward to Elastix, however, in Elastix, can not get my real DID ( in my Asterisk log, showes Executing [s@from-pstn:1] Set("SIP/ABCDE-0000007f", "__FROM_DID=s") in new stack So I guess, I need put application "set“ in my dialplan of my freeswitch, right ? my current public dialplan of my number from SIP trunk provider like this. <extension name="abctel-inbound-1234"> <condition field="destination_number" expression="^(6472221234)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="user/$1"/> </condition> </extension> and in my directory/default, I have a file 6472221234.xml <include> <user id="6472221234"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1234"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="6472221234"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="ABCDE"/> <variable name="effective_caller_id_number" value="6472221234"/> <variable name="outbound_caller_id_name" value="ABCDE"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> What I expect is in Asterisk side, should be my DID from FreeSwitch to Asterisk as "__FROM_DID=6472221234" What's my missing in Dialplan ? or director /default / 6472221234.xml ? in the freeswitch wiki, it said exten to exten need put context = from-internal, and I did this by from-pstn. my Asterisk trunk like this: host=1.1.1.1 username=6472221234 secret=xxxxxxxxxxxxxxx type=friend port=5060 qualify=no insecure=very dtmfmode=rfc2833 deny=0.0.0.0/0.0.0.0 permit=1.1.1.1/255.255.255.255 jbenable=no disallow=all context=from-pstn allow=ulaw trustpid=yes sendrpid=yes register string is : 6472221234:xxxxxxxxxxxx@1.1.1.1 any idea please let me know. -------------------------------------------- My another problem here is the NAT I guess, if my asterisk in my LAN (FreeSwitch still on the public IP. ) the same register shows in my asterisk cli abc*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time 1.1.1.1:5060 N 1009 120 Request Sent can not register on the FreeSwitch. Thank you again. Sincerely, Shaw

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Controlling websockets ip the FS binds to
by Yosi Taguri 17 Jul '14

17 Jul '14

we have machines with two network cards, one is private and one is public. It seems that FS doesn't allow to specify the ip for ws-binding and wss-binding. <param name="wss-binding" value=":7443"/> trying to put an ip before the colon ends up with a profile not being loaded. from looking at the source code it seems that the code looks for sip-ip setting which in our case is set to the internal ip. Am I missing something? -- Yosi Taguri | Founder & CTO | Yalo | http://yaloapp.com <http://ahhhpah.com> | Mobile: +972-54-4816020 | Google Voice US Mobile: +1 (415) 935-1126 | Email: yosi(a)taguri.com

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