September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Per-channel fsctl loglevel?
by Avi Marcus 11 Oct '12

11 Oct '12

Hi. I've been developing a script for someone to handle credit card transactions via IVR. Is there a way to just set the particular channel/call to fsctl loglevel 6, rather than the entire switch..? (dtmf is shown in /log 7) And.. similarly is there a way to blank out the var digits_dialed in the xml_cdr, from within FS, before the end of the call? Thanks! -Avi

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Help needed in a Contact Center solution
by Nitin Tomer 08 Oct '12

08 Oct '12

Hi, I have posted this query earlier, now refining it a bit more. I need to do the following using FreeSwithc - 1. An end-customer calls, the call is handled by mod_ivr. 2. Customer is presented with a menu, she makes selection and also enters some identification. Her call is parked on a pre-decided extension and an entry is made in a database for this call. This entry will hold the selections made by the caller and the extension where call is parked. 3. I will write a service which will poll this database for new calls. 5. New calls will be shown in agents' dashboards. 4. And agent clicks on a query, he is shown an extension where call is parked. He dials that and is connected to the customer. 5. He talks to the customer and resolve his queries. Please guide me on how can do points 1 & 2. Regards Nitin Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL) accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.

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DTMF delay when using FreeSWITCH
by Emrah 04 Oct '12

04 Oct '12

Hi guys, I am comparing this with an Asterisk and FreeSWITCH installation, using the same route, same codecs, same carrier, same phones and same servers… :P I experience a delay when pressing DTMFs on the line that uses FreeSWITCH. I am estimating the delay to be around 500 ms. What are the settings I can fine tune to avoid this? All the best, Emrah

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Help on freeswitch : gtalk-sip conversion
by Sanjay Soni 04 Oct '12

04 Oct '12

Hi, I am trying to get a call working from Gtalk client (Running on my laptop) to the SIP device(Registered as user name 1000). When gtalk client make a call, call is accepted by SIP device but the media is not flowing between Freeswitch and gtalk client. Please note that all the three (Freeswitch machine, gtalk client and SIP device) are in same private LAN inside the firewall. I see that freeswitch is rejecting the local candidate sent by gtalk client and selecting the stun candidate. Why is so ? How can I stop it from doing this. If it could select local candidate I feel call could establish. Thanks Sanjay

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GSMOPEN sms
by Sameer Khan 04 Oct '12

04 Oct '12

Hi, I can't see incoming sms in event plain message I'm using huawei e1550, Is there anything else needed apart from default config?

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GSMOPEN to send long SMS
by Yihui Li 03 Oct '12

03 Oct '12

Hi, I have used mod_gsmopen to send short text messages from my E1550. However, where the length is longer than 160 chars, it always fail. Is there any solution to send long sms like the mobile phone? Thanks, Eric

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snom call pickup blf
by georg＠riseup.net 02 Oct '12

02 Oct '12

Hi, Its been quite a long time, since you started the thread [1]. I've got the same problem you've described. Did you solve this? [1] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/049500…

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 75, Issue 223
by William Alianto 01 Oct '12

01 Oct '12

Hi, I just installed 2 spans of OpenVox's A800E analog card to my server. Here is my freetdm.conf : [general] global_parameter => value [span zt FXO] fxo-channel => 1:1 fxo-channel => 1:2 fxo-channel => 1:3 fxo-channel => 1:4 fxo-channel => 1:5 fxo-channel => 1:6 fxo-channel => 1:7 fxo-channel => 1:8 fxo-channel => 2:1 fxo-channel => 2:2 fxo-channel => 2:3 fxo-channel => 2:4 [span zt FXS] fxs-channel => 2:5 fxs-channel => 2:6 fxs-channel => 2:7 fxs-channel => 2:8 and my freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${moh_uri}"/> </settings>  <analog_spans> <span name="FXS">  <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable_callerid" value="true"/>  </span> <span name="FXO"> <param name="dialplan" value="XML"/> <param name="context" value="default"/>  <param name="tonegroup" value="us"/> <param name="enable_callerid" value="true"/> </span> </analog_spans> </configuration> Also here is my dialplan configuration : <extension name="inbound" > <condition field="context" expression="public" /> <condition field="destination_number" expression="^(\d+)$" > <action application="transfer" data="101 XML default" /> </condition> </extension> <extension name="freetdm.9d" > <condition field="destination_number" expression="^9(\d+)$" > <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="set" data="inherit_codec=true" /> <action application="bridge" data="freetdm/1/a/$1" /> </condition> </extension> But i still cannot make outbound and inbound call using the card. Can anyone help me? Regards

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Pause in bridge
by Systeembeheer 01 Oct '12

01 Oct '12

Hi, I'm trying to configure the following: When a call comes in, it is first bridged to one extension and after 10 seconds to a second extension. It is possible to issue the bridge command with a timeout and continue settings to one extension and a second bridge command to both extensions. There's some problems with that: - The first extension can answer and 'just miss' the first bridge attempt. - If the first call is cancelled by the user, it rings again 10 sec. later. Is there a way to bridge to an extension and bridge to a second extension 10 sec. later without breaking the first bridge? Or bridge to extensions with a pause of 10 sec. for one of them? Thanks! steveN

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Re: [Freeswitch-users] CDR for fail2ban
by Balamurugan Mahendran 01 Oct '12

01 Oct '12

Thanks for the link, Also I wanted to look for outbound numbers and duration to perform some action. Like blocking trail users if they try to use international minutes like more 30 mins or so. Thanks, Bala On Sat, Sep 29, 2012 at 10:05 PM, Avi Marcus <avi(a)avimarcus.net> wrote: > Yes, there is an unmerged mod_fail2ban you can use, which I just doc'ed > for you. > Please update the wiki with more information as you test it. > > Once this has been tested in production, it would be great to merge it > into stable... > > http://wiki.freeswitch.org/wiki/Mod_fail2ban > > -Avi Marcus > BestFone > > On Sat, Sep 29, 2012 at 1:28 PM, Balamurugan Mahendran <b2m(a)a-cti.com>wrote: > >> Is it possible to create separate cdr log file to setup fail2ban. >> >> Thanks, >> Bala >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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