March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Possible mod_smpp
by Michael Collins 08 Nov '12

08 Nov '12

Hello all, If you are interested in giving monetary support to have a professional software firm create mod_smpp then please contact me off list and I will give you more details. Thanks, Michael

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Mod_limit questions
by ivdreg ivdreg 29 Oct '12

29 Oct '12

Hi All, I have some questions regarding mod_limit. 1. I'm using 16674 in production but tested and with 17001. I try mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash never tries to write data to SQL. It works (counts sessions) on log screen but no data in MySQL/Sqlite. It seems that data is cached but cache never flushed to DB. Is anyone has this problem. My limits are: <action application="limit_hash" data="op_${sip_h_X-SRC-Cust} ${destination_number} 5 !INCOMING_CALL_BARRED" /> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust} ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" /> and they are applied on inbound call profile in same extension. 2. I need to implement rule maximum of 50 calls per day/24h from the same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in example "Limit calls to 5 calls every 10 minutes" is about concurrent call but not calls processed as count. Any idea ? Thanks for any help

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Freeswitch in server mode with OpenFire
by krunci 19 Oct '12

19 Oct '12

Hi, I'm new here on this forum, but I have some problem with configuring Freeswitch with OpenFire. First of all, did anybody have succes in configuring Freeswitch with OpenFire, and that it work properly?? Second, I have configure Freeswitch as a component, and add it in Openfire, but when I try to establish a call I have some problems: 1. When I call from SIP to XMPP (openfire) on Sip client have ringing all the time, but on XMPP client side I manage to answer the call, but on SIP side doesn't change anything. (ringing all the time) That means I coundn't establish any call. On Pidgin (XMPP side) is written "Call in progress". 2. In other way, when I try to call SIP client from XMPP client, on XMPP side is trying to call SIP side, but on SIP side doesn't change anything. Like nothing is happening on other side. So if someone who is expert in openfire and Freeswitch can help me with that problems? Thanks Regards, Krunci -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-in-server-mode-wit… Sent from the freeswitch-users mailing list archive at Nabble.com.

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xmpp
by Peter P GMX 19 Oct '12

19 Oct '12

I saw that xmpp is supported in Fresswitch. See wiki: http://wiki.freeswitch.org/wiki/Mod_xmpp_event Has anybody already set this up? I have found no mod_xmpp neither in my mod directory nor in the source? There was also a question: " Q: Is it possible to send commands to fs via xmpp? Answer: Yes. " Anybody knows what can be done here and how to do this? Best regards Peter

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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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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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