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How to receive call using gsmopen
by Anurag Rana 21 Apr '15

21 Apr '15

Hi All, I have micromax usb dongle with a gsm sim in it attached to my system. mod_gsmopen has been compiled and has been loaded. Freeswitch is running. *This is the output of lsusb command -* ---------------------------------------------------------- Bus 002 Device 003: ID 064e:e263 Suyin Corp. Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 006: ID 1c9e:9605 OMEGA TECHNOLOGY Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 011: ID 15d9:0a4e Trust International B.V. Bus 003 Device 010: ID 1a2c:0c21 China Resource Semico Co., Ltd Bus 003 Device 009: ID 1a40:0101 Terminus Technology Inc. 4-Port HUB Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub --------------------------------------------------------------------------------------- *This is the output of dmesg | grep tty command* ---------------------------------------------------------------------- [ 8114.018211] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB0 [ 8114.018840] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB1 [ 8114.019458] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB2 [ 8114.019961] usb 1-1.2: GSM modem (1-port) converter now attached to ttyUSB3 --------------------------------------------------------------------- *This is the content of gsmopen.conf.xml file* *-----*------------------------------------------------------------ <configuration name="gsmopen.conf" description="GSMopen Configuration"> <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="5000"/> </global_settings>  <per_interface_settings> <interface id="1" name="gsm01"> <param name="controldevice_name" value="/dev/ttyUSB3"/> <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>      </interface>  </per_interface_settings> </configuration> ------------------------------------------------------------------------ I searched internet for "how to receive a call using FS" or "How to make an inbound call using gsmopen" but could not find detailed stepwise explanation. Can someone please tell me what I need to do next so that when I call the sim number in dongle, call is received by FS and actions are performed based on dialplan. OR direct me to some link which explains this. (apart from this https://freeswitch.org/confluence/display/FREESWITCH/mod_gsmopen) Thanks Anurag Rana <https://sites.google.com/site/homepagerana/> M.Tech CSE, IIIT-Delhi, https://sites.google.com/site/homepagerana/

4 7

loglevel per caller_id
by Sergey Safarov 21 Apr '15

21 Apr '15

Whether there is a possibility of default logging level to expose NOTICE, and definitely calls from one caller_id set to debug?

4 5

What's wrong with Fax ECM and T.38?
by Victor Chukalovskiy 21 Apr '15

21 Apr '15

Good day, My appologies as this is not specific to FS per say. I'm looking if someone has feedback on a broader topic. I noticed that when T.30 DIS / DCS signals between fax machines negotiate error correction mode (ECM) it often causes issues to various T.38 implementations. I also noticed that certain soft-switches or media gateway would intercept DIS signal and overwrite ECM bit to "false". Thus making far-end fax machine believe call should proceed as non-ECM. In FS there is a channel variable for it. Some Cisco, Metaswitch gear do that as well. Would someone know what is it about ECM Fax mode that makes it less "t.38 friendly"? What forces vendors to implement tricks to prevent Fax machines from negotiating ECM? Many thanks, -Victor

2 1

Dynamic local_stream?
by Ali Pey 21 Apr '15

21 Apr '15

Hello, Is it possible to dynamically set different local_stream files for different users/bridges? For instance I need a different custom MOH for call1 than call2 and they dynamically change as users can upload their own custom moh and I can't change the local_stream.conf.xml config file. Thanks, Ali Pey

4 4

mod_conference CDR format
by Free Switcher 20 Apr '15

20 Apr '15

Hello FreeSwitchers, Does anyone know if it is possible to specify CDR format for conferences? mod_conference outputs XML CDRs by default and I'd like it in CSV format. I can't seem to find any way to specify format anywhere. Thanks for any tips. Regards, Andy

2 1

FreeSWITCH Week in Review (Master Branch) April 11th-17th
by Ken Rice 20 Apr '15

20 Apr '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1JnvNj3 FreeSWITCH Week in Review (Master Branch) April 11th-17th Hello, again. This passed week in the FreeSWITCH master branch we had 14 commits. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. Improvements in build system, cross platform support, and packaging: FS-7060 Added amqp.conf.xml to freeswitch.spec to fix packaging FS-7428 Freeswitch.spec needs to copy mongo driver tarball to libs so that packaging doesn’t fetch from files.freeswitch.org for every build. The following bugs were squashed: FS-7407 Fixed a sip presence issue where the sip proxy changes the transport between the phone and freeswitch and then freeswitch replies back to the proxy over the wrong transport FS-7431 Fixed the parsing of silence streams to be less forgiving of syntax errors FS-7434 Reset jitter buffer when SSRC changes FS-7440 Fix bug in SDP parser, when video a= lines are above the audio a= lines, that could lead to a crash

1 0

skypopen call disconnection after some time ?
by Shabbir abbasi 20 Apr '15

20 Apr '15

this call was disconnected dont know why ?? 2015-04-20 00:27:50.233170 [DEBUG] skypopen_protocol.c:207 [1.4.6~32bit ] [DEBUG_SKYPE 207 ][skype105 ][UP,INPROGRS] READING: |||CALL 5005997 DURATION 602||| skypopen_protocol.c:1082 [1.4.6~32bit ] [DEBUG_SKYPE 1082 ][skype105 ][UP,INPROGRS] CLOSED skypopen_protocol.c:1097 [1.4.6~32bit ] [DEBUG_SKYPE 1097 ][skype105 ][UP,INPROGRS] Skype incoming audio GONE [skype105 ][UP,INPROGRS] len=-1, error: Broken pipe [skype105 ][UP,INC_HNG] Skype outbound audio GONE [skype105 ][UP,INC_HNG] outbound audio server (I am it) EXITING [skype105 ][UP,INC_HNG] incoming audio (read) server (I am it) EXITING [DEBUG_SKYPE 207 ][skype105 ][UP,INC_HNG] READING: |||CALL 5005997 DURATION 603||| [DEBUG_SKYPE 1441 ][skype105 ][UP,INC_HNG] skype call ended any idea ???

3 10

P-asserted-identity in Responses
by Tomáš Boros 20 Apr '15

20 Apr '15

Hi, Is it possible to to enable P-Asserted-Identity in the 18x/200 Responses? I need to copy info from P-asserted identity from Leg-B back to Leg-A. So its like: Invite comes to my freeswitch and the call is bridged to Leg-B. Leg-B answers the call and places a P-Asserted-Identity header into the 200 OK response 200 OK response on the Leg-A should containt P-Asserted-Identity of Leg-B (callee). Thank you, -- Tomáš Boros

1 0

RTP packet loss on outgoing streams
by mbo 20 Apr '15

20 Apr '15

Hello, we need to monitor the voice quality on our server. To test this, we setup a test environment with 2 freeswitches, originating calls from freeswitch 1 and then answer them on freeswitch 2 and play an announcement. To analyze the traffic, we us tshark with the following command: tshark -q -f 'udp' -o rtp.heuristic_rtp:TRUE -z rtp,strems -w test-pcap The result is, that we see packet loss around 20% on outgoing RTP streams, when we have more then 20 calls. The load average shown with the top command is always < 0.3 with a 4 core CPU and CPU load is less then 10%. ========================= RTP Streams ======================== Src IP addr Port Dest IP addr Port SSRC Payload Pkts Lost Max Delta(ms) Max Jitter(ms) Mean Jitter(ms) Problems? 192.168.1.134 27590 192.168.1.135 22576 0xCD7A189D ITU-T G.722 800 2 (0.2%) 59.93 0.11 0.05 X 192.168.1.135 22576 192.168.1.134 27590 0x5D4E50CD ITU-T G.722 599 202 (25.2%) 160.24 0.16 0.05 X 192.168.1.134 28172 192.168.1.135 29704 0xD53EC9AD ITU-T G.722 798 2 (0.2%) 59.99 0.07 0.04 X 192.168.1.135 29704 192.168.1.134 28172 0x5D50942D ITU-T G.722 713 86 (10.8%) 199.98 0.08 0.04 X 192.168.1.135 25794 192.168.1.134 16460 0x5D51E43D ITU-T G.722 633 165 (20.7%) 180.25 0.75 0.07 X Instead of tshark, we also tried to capture SIP & RTP with tcpdump with the following command: tcpdump -i any -T rtp -vvvvv -s 0 -w test.pcap If we analyze this file with wireshark (TELEPHONY->RTP->Show all Streams), it shows almost always 0% for packet loss. Only if we increase the load to 50 calls we can see a similar scenario where inbound calls still have < 1% packet loss but outgoing RTP streams have up to 20 % packet loss. We also repeated the same test with freeswitch 1.4.15 with much more powerful hardware with almost the same results. The question is what do we see here. Does tshark show us wrong result? Can we more trust on the results captured with tcpdump or is there finally an issue in freeswitch when sending outgoing packets? Or is the method to capture on the server in general the wrong approach? Thanks for any comments and hints Markus

3 7

auth_app fields from the xml_radius config module
by Густаво Силва 20 Apr '15

20 Apr '15

How can I configure the fields in <auth_app> part in the mod_xml_radius.config.xml file? I have set like this: <auth_app> <connection name="testing"> <param name="authserver" value="127.0.0.1:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/src/freeswitch/src/mod/xml_int/mod_xml_radius/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_network_ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" format="dst-number-in=%s" /> <param name="Called-Station-Id" variable="sip_to_user" format="%s"/> <param name="Calling-Station-Id" variable="sip_from_user" format="%s"/> <param name="User-Name" variable="sip_from_user" format="%s"/> </fields> </auth_app> I got this log in my radius server: Mon Apr 20 13:15:13 2015 : Auth: Login OK: [1008/<via Auth-Type = DIGEST>] (from client localhost port 0 cli 1008) Mon Apr 20 13:15:13 2015 : Info: rlm_radutmp: Login entry for NAS localhost port 0 duplicate Mon Apr 20 13:15:13 2015 : Auth: Login incorrect: [1008/<no User-Password attribute>] (from client localhost port 0 cli 1008) My goal is to authenticate a call. Regards, Gustavo

1 0