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Puzzling DTMF problem
by Mario G 24 Jan '16

24 Jan '16

Called an automated system. The first couple of prompts work (press 1 for english, etc.). Then DTMF is not recognized. For a few places I notice none of the DTMF is recognized. But it’s really a puzzle when it’s partial for a the same called number. I always thought RFC2833 is what should be set. Any thoughts? Mario G

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Freeswitch 1.6.6 crash when a call returns from HOLD
by Yehavi Bourvine 24 Jan '16

24 Jan '16

Hello, After trying to upgrade to version 1.6.6 I've found that quite often Freeswitch crashes when a user tries to un-hold a held call. I've already opened FS-8739 a few days ago, but no comment yet. With version 1.6.5 it works ok. Has anyone also noticed it? I tried it with Polycom phone, as this is what I have in my lab. Thanks, __Yehavi:

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faxes send bad
by Денис 24 Jan '16

24 Jan '16

Hello. We want migrate from asterisk to FS and now we need stable faxing, but with aster - 90% faxes success, with FS - only 20%. disable_v17=true is set, but usually we got Disconnected after permitted retries What i can tune and how? bad: https://pastebin.freeswitch.org/24532 success transmit: https://pastebin.freeswitch.org/24533

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Learning FreeSWITCH - Can't call into test installation
by William Colls 23 Jan '16

23 Jan '16

I am new to freeSWITCH, and working trough the examples from the FreeSWITCH (1.2) Book. I have the system built, and running, and everything looks OK. The system registers with my VoIP provider (voip.ms), an IP phone registers with FreeSWITCH, and I can call out to the POTS. However, when I try to call into the IP phone, I can see the incoming call information at fs_cli, but it instantly hangs up, and never goes through to the intended destination set up in dialplan/public/01_voipms.xml. The message that I get is [NOTICE] switch_core_state_machine.c:315 sofia/external/613XXXXXXX(a)158.85.70.XXX has executed the last dialplan instruction, hanging up Appreciate any pointers as to what/where I should be looking for the problem. Thanks for your time. William.

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Identify billable calls
by John Nash 23 Jan '16

23 Jan '16

I am using bridge application to connect to carrier gateways and using CDR csv module for billing. In my logic I am only billing calls which have billsec>0 now I wonder if this is correct way to find what calls are connected I mean may be some calls I am ignoring which were connected for less than 1 second?..Is that possible?...what is the sure shot way to filter connected calls?

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mod_callCenter scalability
by Varsha Agarwal 22 Jan '16

22 Jan '16

Hi, Does anyone have any data on how mod_callCenter scales? How many queues, concurrent calls, number of agents can be supported on one instance of FreeSwitch? I looked through the whole documentation but did not find any data. Thanks, Varsha

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Linux clocksource TSC vs HPET
by François 22 Jan '16

22 Jan '16

Hi all, I'm trying to find a solution for an "RT throttling" issue under relatively high load (20cps) on linux 3.10 and CPU Xeon E5-1650v2 (6cores). A similar jira ticket FS-8620 has a recommendation to set the clocksource to "hpet": echo hpet > /sys/devices/system/clocksource/clocksource0/current_clocksource I read somewhere that the default clocksource (TSC) is lighter in CPU usage because it only requires reading a register that is synchronized across all cores... Considering FreeSWITCH needs: -> Is HPET recommended over the default "TSC" clock? -> Does a change in clocksource have immediate effects or does it require an FS restart/reboot? -> Also, is it recommended to disable RT-throttling to resolve these types of issues? echo -1 >/proc/sys/kernel/sched_rt_runtime_us echo -1 >/proc/sys/kernel/sched_rt_period_us Thanks, François.

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Possible performance problem
by Jose Fco. Irles Durá 22 Jan '16

22 Jan '16

Hi, I have a FreeSWITCH (1.4.26) for click2call call center application in production. Some times, I have detected some delay on bridge operations (captured with tcpdump): 1. originate call to softphone and softphone answers 2. deflect call 3. bridge to a gateway in the step 3, some times (I suspect in peak hours) when the FreeSWITCH receives de 200 OK from the upstream gateway when de called party answers, it delays until 3 seconds to send the 200 OK to the softphone leg. How I can debug if it is a performance problem? The system is a lxc container on top of drbd disk. On peak hours the system has around 60 concurrent calls The calls are recorded in ram disk The FreeSWITCH is connected to localhost postgresql database for its core db. Best regards

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How to transfer call from one extension to another extension ?
by ankitdoshi 22 Jan '16

22 Jan '16

Hi All, Anyone please help me out for transfer a call ? let me give you an example : i have 3 extension 1- caller(1000) 2- receiver(1001) 3- transfer(1002) I am dialing from 1000 and call is received by 1001 then push transfer button and press 1002 for transfer but its not going to transfer.I have set below parameters in sip profile but still its not going to work 1. manual-redirect=true,2. disable-transfer =false Can anyone please help me out for this ? Any suggestion will be appreciated . ----- Thanks Ankit Doshi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-transfer-call-from-one… Sent from the freeswitch-users mailing list archive at Nabble.com.

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the unit for answered_time
by ik 22 Jan '16

22 Jan '16

Hello, I require to calculate on hangup the duration of a call regardless of the CDR, and do some logic if it's above a range. I'm placing at the api_hangup_hook a cli parameters that one of them is answered_time. I need the data in minutes unit, and the data seems to be a timestamp in microseconds, but converting it to seconds (e.g. answered_time / 1000000) provides me a very low range of number, sometimes even less then a millisecond. I can't seems to find at the documentation what is the unit for the answered_time in order to properly convert it to my requirements. I'm also missing the place at the source code where the data is set to the variable (here is the grep result (at the src directory): grep answered_time ./* -R ./mod/event_handlers/mod_cdr_mongodb/mod_cdr_mongodb.c: bson_append_long(&cdr, "answered_time", caller_profile->times->answered); ./mod/languages/mod_managed/managed/ChannelVariables.cs: const string answered_time = "answered_time"; ./mod/languages/mod_managed/managed/ChannelVariables.cs: get { return readUsecsDateTime(answered_time); } ./switch_caller.c: if (!strcasecmp(name, "answered_time")) { ./switch_core_session.c: } else if (!strcmp(tag3->name, "answered_time")) { ./switch_ivr.c: if (!(time_tag = switch_xml_add_child_d(x_times, "answered_time", t_off++))) { ./switch_ivr.c: cJSON_AddItemToObject(j_times, "answered_time", cJSON_CreateString(tmp)); So what am I missing here? Thank you Ido

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