July 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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Saving phone information from a Polycom IP501
by Todd Bailey 27 Sep '12

27 Sep '12

Just wondering if it's possible to download data from a phone and save it into separate files or into a database? I just save a number into the phone and it occurred to me the minute I have to reboot or power cycle the phone, all contact info will be lost. If possible, I'd like to store the contact information and call history into a database for future reference, and perhaps being able to store that information into my email and contact management application (currently Evolution)

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FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up
by Avi Marcus 26 Sep '12

26 Sep '12

Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.) I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help. I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel. Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA Specifically: 1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up? 2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too. Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files? And anything else I may be overlooking. Thanks guys! -Avi Marcus

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Timezone with time conditions
by Spencer Thomason 24 Sep '12

24 Sep '12

Hello all, Has anyone found an elegant solution to using time conditions with multiple locations? For example suppose you have a multi tenant system with time conditions, is there any way to set the call time zone on a per call basis? Thanks, Spencer

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mod_dingaling picking wrong IP address / no audio?
by Mark Campbell-Smith 14 Sep '12

14 Sep '12

Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks!

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mod_dingaling & inbound audio
by Locutis of Borg 14 Sep '12

14 Sep '12

Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio. FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem In and out calling on SIP to provider works fine in both cases. So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users? Thank you for you help. "Resistance is Fruitful"

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Variable continue on fail
by Виталий Давудов 13 Sep '12

13 Sep '12

Hi all! In my dialplan I've included variable continue on fail: <extension name="Dial Out SIP-M200 Mobile numbers"> <condition field="destination_number" expression="^(89\d{9})$"> <action application="export" data="t38_passthru=true"/> <action application="set" data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> <action application="info"/> <action application="bridge" data="sofia/gateway/SIP_M200/$1"/> <action application="limit_execute" data="hash limit SMG-1016M 60 bridge sofia/gateway/SMG-1016M/$1"/> </condition> </extension> And if FS recieve from first gateway USER_BUSY, then FS try to bridge this call to another gateway. Although in line <action application="set" data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> there is no code Q.850 = 17. How resolve this issue? -- Best regards, Vitaly Davudov "VIP-TELECOM-SERVICE" Ltd. ("ETERIA" Group of companies) http://www.vts24.ru

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freeswitch - sangoma CAS T1 - a101 configuration
by copycall 07 Sep '12

07 Sep '12

hi, i'm having trouble getting instructions to configure freeswitch with a sangoma a101 T1 card for CAS, B8ZS, ESF. the first application i seek to solve is using this setup as a gateway (replacing a dialogic dmg 2000) for sip from my itsp to the tdm endpoint: a legacy centigram - mitel voicemail server. any help will be greatly appreciated. thank you, dave

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How to set a variable for a call going from gateway?
by "Гришин В.С." 04 Sep '12

04 Sep '12

After I read a http://wiki.freeswitch.com/wiki/Sofia.conf.xml#Variables I configured FS [root@freeswitch1 external]# more /usr/local/freeswitch/conf/sip_profiles/external/SMG-1016M.xml </include> <gateway name="SMG-1016M"> <param name="realm" value="10.10.10.1"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="25"/> <variables> <variable name="SMG-1016M_inbound_var_name" data="this_is_inbound_call" direction="inbound"/> <variable name="SMG-1016M_outbound_var_name" data="this_is_outbound_call" direction="outbound"/> <variable name="SMG-1016M_both_var_name" data="this_on_any_direction_call"/> </variables> </gateway> </include> [root@freeswitch1 external]# I insert the info application into public context of dialplanbefore a transfer application? make call but don't see variables (SMG-1016M_inbound_var_name,SMG-1016M_outbound_var_name,SMG-1016M_both_var_name) in fs_cli. I want to see variable = <gateway name="SMG-1016M"> into dialplan for each gateway in (single) external sip profile. How it to make? Vladislav Grishin

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Open Source unified autodialer software released
by tahir almas 28 Aug '12

28 Aug '12

Pleased to announce the release of open source Fax , SMS and Voice broadcasting software solution ICTDialer http://www.ictdialer.org developed over reknown Drupal Conent Mnagment System and powerfull Plivo Communication framework , Your contribution and suggestions are welcome Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation.

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