June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hong Kong/Singapore VoIP Carriers
by Eric Detterman 02 Aug '14

02 Aug '14

We're planning for a FS cluster in Singapore and Hong Kong and are looking for carriers to integrate with - preferably ones with higher quality & service tiers, even if it means paying a bit more. Does anyone have any recommendations for companies like this? So far Google hasnt been overly helpful.

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e911
by Joli Martinez 17 Jul '14

17 Jul '14

Hello I am looking for an open source e911 service where I can manage multiple locations centrally. Does anyone have any experience with any software that can do this? Thanks,

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Re: [Freeswitch-users] condition
by Andrew 07 Jul '14

07 Jul '14

What I am trying to do is wait for the variable to be set. But it seems that the variable is never set in this code. Anyone have an idea on how to wait for a variable to be set? <extension name="public_extensions"> <condition field="destination_number" expression="^.*$"> <action application="set" data="hangup_after_bridge=true"/> <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="20"/> </condition> <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="500"/> </condition> <action application="bridge" data="${my_dialstring}"/> </condition> </extension> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of aademattia(a)comcast.net Sent: Saturday, June 28, 2014 2:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] condition I am trying to find out if the variable is not set. The problem is this condition fires if the variable is set or not set. If the variable is not set, I want it to sleep. Sent from my HTC ----- Reply message ----- From: "Ken Rice" <krice(a)freeswitch.org <mailto:krice@freeswitch.org> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > Subject: [Freeswitch-users] condition Date: Sat, Jun 28, 2014 1:24 PM thats tesing for an empty string... test to see if what you expect to be there and if not use anti-action to sleep Ken Sent from my iPad On Jun 28, 2014, at 11:15, "Andrew" <aademattia(a)comcast.net <mailto:aademattia@comcast.net> > wrote: Hi, I want to see if the variable is set. If its not I want to sleep for 500 mill sec. This does not seem to work. <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="500"/> </condition> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@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 <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org

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python+ESL
by Денис 07 Jul '14

07 Jul '14

who wrote event handling with ESL.py? Code: #!/usr/bin/python import ESL # Establish an inbound connection to FreeSWITCH con = ESL.ESLconnection("localhost","8021","ClueCon") print con.connected() print con.getInfo() # set event command to receive all events con.events("plain", "all"); #while(1): # or whileon.connected()): while(con.connected()): e = con.recvEvent() if e: print "Event: ",e.serialize() Out: 0 None run fs_cli: # fs_cli -H 127.0.0.1 -P 8021 -p ClueCon -b ... +OK log level [7] Where exceptions with error?How to properly connect?

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mod_spandsp: Inband DTMF not processed
by Nagalenoj H. 04 Jul '14

04 Jul '14

Hi, I am facing an issue with using Inband DTMFs. Inband DTMFs are getting detected in switch_rtp.c but then it is not processed further by mod_spandsp_dsp.c. This issue is inconsistent, for few calls it is working and not working for few other calls. For a working call, I'm getting the logs as below, 2014-06-19 21:24:40.334513 [DEBUG] switch_ivr_play_say.c:1682 done playing file file_string:///usr/local/freeswitch/sounds/music/switch/2539_SB_welcome_switch.wav!/usr/local/freeswitch/sounds/music/switch/1sec_silence.wav!/usr/local/freeswitch/sounds/music/switch/SB_landing_IVR.wav 2014-06-19 21:24:44.114522 [DEBUG] switch_rtp.c:3793 RTP RECV DTMF 3:1280 2014-06-19 21:24:44.694512 [DEBUG] mod_spandsp_dsp.c:372 DTMF BEGIN DETECTED: [3] 2014-06-19 21:24:44.854513 [DEBUG] mod_spandsp_dsp.c:384 DTMF END DETECTED: [3], duration = 165 ms 2014-06-19 21:24:45.434531 [DEBUG] switch_rtp.c:3793 RTP RECV DTMF 9:1280 2014-06-19 21:24:45.994513 [DEBUG] mod_spandsp_dsp.c:372 DTMF BEGIN DETECTED: [9] 2014-06-19 21:24:46.154514 [DEBUG] mod_spandsp_dsp.c:384 DTMF END DETECTED: [9], duration = 165 ms 2014-06-19 21:24:47.434530 [DEBUG] switch_rtp.c:3793 RTP RECV DTMF 2:1280 2014-06-19 21:24:48.034523 [DEBUG] mod_spandsp_dsp.c:372 DTMF BEGIN DETECTED: [2] 2014-06-19 21:24:48.034523 [CRIT] sb_main.js:390 digits pressed after collect input: 392 On the failure calls, I'm getting only the RTP RECV DTMF log and not the DTMF BEGIN and DTMF END DETECTED log messages and because of this, dialplan application is not receiving any DTMFs. I've attached the pcap file and full fs log for failure call. What could be the cause of this inconsistent behavior? Kindly help me to rectify this. -- Regards, Nagalenoj H.

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No MWI notification with TLS
by Torsten Hantzsche 04 Jul '14

04 Jul '14

Hi, can somebody confirm this effect? - FreeSWITCH version: 1.5.13b on FreeBSD - CA, agent.pem and cafile.pem were created by gentls_cert and the internal profile was configured to enable TLS with only SSL23 cipher (according to https://wiki.freeswitch.org/wiki/SIP_TLS) - a user has at least one waiting voice mail - my UAC is CSipSimple_m0-18/r2423 a connection via UDP shows the following flow (UAC - UAS): ---> REGISTER <--- 401 Unauthorized ---> REGISTER with credentials <--- 200 OK ---> SUBSCRIBE <--- 202 Accepted <--- NOTIFY with payload "Messages-Waiting: yes" and "Voice-Message: 2/0" ---> 200 OK Here CSIPSimple correctly indicates waiting messages. However, if i connect the same user via TLS (SSL23), i get only this flow: ---> REGISTER <--- 401 Unauthorized ---> REGISTER with credentials <--- 200 OK ---> SUBSCRIBE <--- 202 Accepted And that's all, no NOTIFY is sent from FS and CSIPSimple shows no MWI-Message. Is there possibly more to configure to work the same way with TLS? Thanks for any help and hints Torsten

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RTCP and OPUS
by Kamil Nigmatullin 01 Jul '14

01 Jul '14

Hello all. What I basicly want is to have real adaptive codec on both sides (client and FS), that could exchange link quality information through RTCP. As far as I see, after a brief looking through some parts of source code of FS, mod_opus works as static codec and don't change its behavior when Internet link is changing because it has no communication with RTCP module. Could anybody of developers recommend me the most efficient way to deploy this communication process through Freeswitch Core? Is there a way to export and import information from/to module/core during the call? Thanks in advance.

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Problems with RTP delay at around 300 concurrent calls
by D D 30 Jun '14

30 Jun '14

Hi, I would like to put 500 - 1000 calls into a FS system. I have a Dell R900 server with these properties: 8 processors at 2.40GHz 8GB RAM Gigabit ethernet I use sipp to send RTP streams. I use the FS settings recommended here: https://wiki.freeswitch.org/wiki/Performance_testing_and_configurations including the in-memory core.db. I also have used the "-rp" and "-nosql" settings on the FS command-line, and also used the "chrt" command to increase the priority to real-time. In all cases, at around 300 concurrent calls, the RTP inter-packet delay becomes unacceptably high. The voice quality is severely impacted. During the test, the CPU is not overloaded. It is using only about 1.5 of the 8 processors. Are these results expected? Has anyone gone higher than 300 concurrent calls? Is there anything else that I can do with configuration to improve throughput? Thanks, David

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send_dtmf not working
by Matt Broad 30 Jun '14

30 Jun '14

Hi, I have the need to send a dtmf tone, but send_dtmf does not seem to work. I have the following in a dialplan, but doing a sip trace the dtmf digits are not sent. I have tried removing the start_dtmf and start_dtmf_generate, but still no joy. <action application="start_dtmf" /> <action application="start_dtmf_generate"/> <action application="answer"/> <action application="sleep" data="2000"/> <action application="send_dtmf" data="0123456789ABCD*#@100"/> <action application="sleep" data="2000"/> I am using FreeSWITCH Version 1.4.6+git~20140603T193516Z~9479729340~64bit (git 9479729 2014-06-03 19:35:16Z 64bit) thanks Matt

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P-Asserted-Identity header issue
by Ričardas Stoma 30 Jun '14

30 Jun '14

Hello guys, i have a problem when i try to read P-Asserted-Identity header. Let's say i send to freeswitch this header: P-Asserted-Identity: <sip:123456789@192.168.0.73:5060;user=phone> But when i try to read this header in my dialplan, i get: 123456789 When i check channel variables i see: variable_sip_P-Asserted-Identity: 370370370370 maybe there is a way to read full P-Asserted-Identity header? I need to resend this header without modifying it. Is it possible?

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