- FreeSWITCH-users - FreeSWITCH Mailing Lists

Optimal way to monitor call stats
by Muhammad Naseer Bhatti 21 Feb '12

21 Feb '12

Hi, I was trying to figure out what should be the optimal way to generate live stats to monitor calls. For example, connect to event_socket, subscribe to CHANNEL_ANSWER, CHANNEL_PROGRESS, CHANNEL_DESTROY etc .. and parse them accordingly? Or if there is anything better than this to handle. Since the switch is generating higher number of calls per second, if we opt to this method won't this be too heavy for the server resources? Thanks, Goni

2 1

LOSE_RACE, NORMAL_CLEARING and billsec==0
by Joe Flemmings 21 Feb '12

21 Feb '12

When I call a ring-group and answer the call, both the LOSE_RACE and NORMAL_CLEARING channels have a billsec==0 in the CDR How can I fix this to get the real time on the channel with NORMAL_CLEARING? I'm using cdr csv

2 2

COLP using To: Header
by Markus Lindenberg 21 Feb '12

21 Feb '12

Hi everybody, it seems some phones (Snom at least) support COLP (connected line update) in ringing phase by having the To:-header changed in the 180 ringing to a-pty. Snom have a nice detailed illustration on their wiki: http://wiki.snom.com/Category:Connected_Line_Identification_Presentation_(C… So if Fritz (111) calls Hans (222): INVITE sip:222@myshop From: "Fritz" <sip:111@myshop>... To: <sip:222@myshop>... The response sent to 111 should look something like this: SIP/2.0 180 Ringing From: "Fritz" <sip:111@myshop>... To: "Hans" <sip:222@myshop>... So basically i suppose i want freeswitch to stuff a effective_caller_id_name (if set) into the To: header in the a-leg, given somthing like the following dialplan snippet: <action application="export" data="effective_callee_id_number=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_number)}"/> <action application="export" data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> Is this possible? Thanks, Markus

1 0

FreeSWITCH Security Hardening Project
by Brian West 21 Feb '12

21 Feb '12

While sitting here in the hospital I have been talking to Philip Zimermann (ZRTP) via phone... And we have talked about a FreeSWITCH hardening project (NO JOKES PLEASE). This is taking FreeSWITCH and security to the MAX. I would like to start this project to make FreeSWITCH more secure in the long run. (not saying that its not secure now). Who would be interested in such a project? Thanks, -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org http://www.freeswitch.org

9 15

Latest GIT build error for mod_dptools
by afshin afzali 21 Feb '12

21 Feb '12

Hi, In latest freeswitch in mod_dptools.c function audio_bridge_function there is unused "channel" variable which causes build error! BEST, -- afshin

2 1

Virtualized Freeswitch?
by Tim Nelson 21 Feb '12

21 Feb '12

Greetings all- I'm quite experienced with Asterisk, and specifically, usage in a virtualized environment. It works great with a bit of tuning. I'm starting to work more with Freeswitch, and as such, I'm looking to implement it in a virtualized environment. My initial installation was onto a CentOS 6 OpenVZ container. However, the logs are showing this (many times): 2012-02-20 18:43:51.846882 [WARNING] switch_scheduler.c:114 Task was executed late by 280 seconds 1 heartbeat (core) 2012-02-20 18:43:51.846882 [WARNING] switch_scheduler.c:114 Task was executed late by 260 seconds 2 check_ip (core) 2012-02-20 18:46:21.496538 [CRIT] switch_time.c:798 Virtual Migration Detected! Syncing Clock 2012-02-20 18:46:21.496538 [CRIT] switch_time.c:810 If you see this message many times try setting the param enable-clock-nanosleep to true in switch.conf.xml or consider a nicer machine to run me on. I AM *FREE* afterall. I completely understand the issue with realtime applications such as VoIP in a virtualized environment with limited timing facilities. However, OpenVZ makes use of a shared kernel which in my prior experience, does not have the issues associated with fully virtualized technologies like VMWare, KVM, Xen, Hyper-V, etc. Am I incorrect or is this issue specific to Freeswitch (but apparently not so in use with Asterisk) ? There are a couple of wiki entries [1][2] which seem to indicate others are using Freeswitch with OpenVZ with success. Am I mistaken? Looking for some pointers if you can lend some insight into this issue. Maybe the errors are harmless? I have not actually pushed any traffic through this installation yet, due to the errors. Thanks! --Tim [1] http://wiki.freeswitch.org/wiki/FreeSWITCH_in_OpenVZ [2] http://wiki.freeswitch.org/index.php?title=Virtualization&redirect=no

7 12

FS protocol interoperability
by Josh 21 Feb '12

21 Feb '12

Very simple query: If I configure my (external) gateway to use SRTP/ZRTP/SKYPE and then configure all my internal clients to register to FS using "regular" (understand unsecured) sip, would it be possible to route (i.e make and receive) calls between the gateway and the internally registered clients? If so, what are the likely problems I could face? Many thanks!

3 4

CDR mysql
by Miha Zoubek 21 Feb '12

21 Feb '12

Hi, one of the user on #freeswitch told me, that the new modul will come out for mysql in cdr. Any news about it? Thansk! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

2 2

How to increse max session of freeswitch
by virendra bhati 21 Feb '12

21 Feb '12

Hi List, I have installed Freeswitch with the help of vBiling. But when i checked that status then i saw that only 200max session i can used. Any one can help me how I can increase these session to 1000 or more then that. Below is the status of my freeswitch freeswitch@internal> status UP 0 years, 4 days, 21 hours, 26 minutes, 4 seconds, 380 milliseconds, 632 microseconds FreeSWITCH is ready 46 session(s) since startup 0 session(s) 0/30 200 session(s) max min idle cpu 0.00/38.00 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2

2 1

Oddities with Polycom's vesion 4.0.0
by Yehavi Bourvine 21 Feb '12

21 Feb '12

Hello, I've upgraded our Polycom phones from firmware 3.3.2 to 4.0.0 and have the following oddities: - On shared lines the call lists are "mixed": - No calls are in outgoing calls. - answered incoming and outgoing calls are placed in incoming calls. - unanswered incoming and outgoing calls are placed in the answered calls. - When the destination is busy there is no busy sign on the screen - the call is disconnected immediately and the phone returns to an idle state. I've opened today a support call with Polycom, and will update when I get replies. Regards, __Yehavi:

2 3