February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

IP Failover with Sofia Recover - not recovering?!
by Avi Marcus 15 Mar '11

15 Mar '11

I finally figured out how to get pacemaker to start profiles/trigger a sofia recover on moving the floating IP (but an FS crash it still can't handle) Anyway - I 1) set up odbc for the core, and mysql master-master replication, so I see calls & channels being logged in both 2) I set track calls in all the profiles So when I set it to transfer from one node to another, I'm kinda surprised I'm not getting a recovery of the calls. I do see that the DB has a hostname stored - "sip1" or "sip2" depending on where the call originated, but even if I change it manually, a sofia recover seems to ways say "no calls to recover" Am I missing something?! -Avi Marcus

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Re: [Freeswitch-users] mod_callcenter features
by Rob Hutton 14 Mar '11

14 Mar '11

1. Try something like: reloadxml callcenter_config queue load|reload queue_name 2. I don't believe so. Hit Moc on the IRC channel. I am trying, also... -- Thanks, Rob On Friday 31 December 2010 00:32:46 Rajkumar K wrote: > Hi, > > 1. In mod_callcenter, there is no command like "callcenter_config queue add" > for adding queues dynamically. We need to configure it in > callcenter.conf.xml file and run "callcenter_config queue reload <queue > name>". So is there any other ways to add queues. > > 2. The mod_callcenter automatically bridges the call once it identifies the > available agent. But the requirement is to identify the available agent and > leave the bridge control to our dial plan. There we need to other operations > (like announcing caller name to agent) instead of bridging the call. > > regards > rajkumar k > > On Thu, Dec 30, 2010 at 6:12 PM, Rob Hutton <justlikeef(a)gmail.com> wrote: > > > Some of the API commands are on in the Wiki under the heading API: > > > > http://wiki.freeswitch.org/wiki/Mod_callcenter > > > > Everything starts with callcenter_config, so you can feel your way around > > the command line and everything is fairly self explanatory. > > > > What do you mean control the call flow? Can you give an example of what > > you are trying to accomplish? > > > > -- > > Thanks, > > Rob > > On Monday 27 December 2010 04:42:29 rajkumar wrote: > > > > > > Hi, > > > > > > I am developing an application with mod_callcenter. I need to know the > > > following about mod_callcenter. > > > > > > * Is it possible to add/update/delete the queue configurations > > dynamically > > > without using static xml configurations. > > > * How can I control the call flow (for playback message and recording) > > > before and after bridging the call. > > > > > > Thanks in advance > > > > > > regards > > > rajkumar k > > > > > >

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exec app on originate
by Madovsky 14 Mar '11

14 Mar '11

when I do this : originate /sofia/gateway/blabla/9999999999 &app(hahaha) is the ap executed at early media or after answer ? Thanks

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freeswitch -nc -nosql
by mazilo 12 Mar '11

12 Mar '11

It had been a while I didn't get a chance to upgrade my FS git until today. With a newly built FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25 21-06-22 +0100), at least a new problem surfaces where FS Segmentation fault as shown below: 2011-02-25 21:30:34.277865 [CONSOLE] switch_core.c:1764 _____ ______ _____ _____ ____ _ _ | ___| __ ___ ___/ ___\ \ / /_ _|_ _/ ___| | | | | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | | | || | | |_| | | _|| | | __/ __/___) |\ V V / | | | || |___| _ | |_| |_| \___|\___|____/ \_/\_/ |___| |_| \____|_| |_| ************************************************************ * Anthony Minessale II, Michael Jerris, Brian West, Others * * FreeSWITCH (http://www.freeswitch.org) * * Paypal Donations Appreciated: paypal(a)freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/ * ************************************************************ 2011-02-25 21:30:34.278007 [CONSOLE] switch_core.c:1767 FreeSWITCH Version 1.0.head (git-0f7682a 2011-02-25 21-06-22 +0100) Started. Max Sessions[1000] Session Rate[30] SQL [Enabled] Segmentation fault @DockStar:/# If I add a -nosql switch, i.e. freeswitch -nosql, it runs OK but it keeps dumping the following message on the CLI: 2011-02-25 21:36:19.378416 [ERR] switch_core_sqldb.c:1721 Error Opening DB! So, whatI do I need to tweak to make the above message go away? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-nc-nosql-tp6066718… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Call back from voice mail creates loop
by Tim St. Pierre 11 Mar '11

11 Mar '11

Hello, I'm trying to set up the option in voicemail so that users can return a call directly from VM. In my voicemail.conf.xml profile I have specified the dialplan as XML and the context as "internal", which is what I use for calls coming from our endpoints or authenticated users. In our internal dialplan, we do some pattern matching and call an appropriate lua dialplan script to do build a set of instructions. Problem is, the lua scripts are looking at the destination_number variable to decide where to send the call. This variable is still set to "voicemail" from when the user called into their voicemail. Shouldn't the voicemail app have set that to the number we want to call back? I used the info app to look at the variables, and I didn't see anything obvious. Is destination_number the correct variable to use in a lua script that generates a dialplan? -Tim

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British sounds
by Alex Crow 11 Mar '11

11 Mar '11

Does anyone know of either of the following: 1. A source (paid or free) of British English sounds (preferably female voice) for Freeswitch, preferably covering all the same sounds as those supplied. or 2. A script to convert and rename sounds from an Asterisk install to FS? We already paid for British female sounds for Asterisk so if conversion will cover at least voicemail prompts it would be a start. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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mod callcenter
by Sam 10 Mar '11

10 Mar '11

Hello, Was testing callcenter module and found out that at times it gives error " invalid application callcenter " and after reloading the module it works fine. Some time also happens that if I reload the module it do not reloads the parameters of the callcenter like the agents & tires. It just unload & reloads even if there are changes to the specifications of agents. Regds Sam

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Virtualisation and freeswitch
by Grygoriy Dobrovolskyy 08 Mar '11

08 Mar '11

Hello list, le me introduce myself first, I am Grigoriy Dobrovolskyy currently working as voip administrator and call centeres integrator. I worked a lot with asterisk, i think Fs is a great software but not easy to learn. I have googled my question and did not found a clear answer. With what technology i can virtualise freeswitch without timer problems ? I know that i can use OpenVZ but it's more 'jail' then virtualisation. I heard that Amazon EC with High CPU instance works (confirm?) Can someone give me info of a succesfull installations with Conferences, Moh / mixing ? And a technology used ? Thank you.

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fs_cli socket connection error
by deniro 04 Mar '11

04 Mar '11

What would be possible reasons for this and how to resolve? running fs 106 on ubuntu 10.04 server /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021 [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error] was working fine before I installed mod_xml_cdr configure --prefix=/opt/freeswitch --without-libcurl make mod_xml_cdr-install (no errors) in modules.conf.xml <load module="mod_event_socket"/> is there a packaged version of mod_xml_cdr (for fs 1.0.6)? thx deniro--

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Freeswtich as a media proxy between ipv4<=>ipv6 using Polycom HDX8006 SIP UA-s
by MÉSZÁROS Mihály 03 Mar '11

03 Mar '11

Hi, 1. Is it possible to create ipv6 <=> ipv4 media proxy from FreeSwitch? So i need to mangle SDP to replace ipv6 to ipv4 and vice-versa. (but i use fnacy things like BFCP,FECC(H.224),secondary video) 2. Further more I need to know that FreeSwitch can function as a real media proxy? So can it PROXY as MEDIA like BFCP(TCP!), FECC and secondary video? 3. Can i use more than one stream so more than 1 audio + 1 video stream in a sip call in proxy media mode? For example 1 audio + 2 video stream (people+presentation) Example SDP piece for BFCP, and FECC(H.224): m=application 49158 RTP/SAVP 100 a=rtpmap:100 H224/4800 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lv6Y5eO/RSDDOFNvbFmM0q9tfrt3/ZxXqPJx4Pj5|2^31 m=application 0 TCP/BFCP * a=floorctrl:c-s a=setup:actpass a=connection:new Any help highly appreciated! Thanks, Misi

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