March 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Crashed- Segmentation Fault
by Achintha 03 Apr '18

03 Apr '18

Hi all, Our freeswitch server handles 3000 – 4000 calls per day. We have custom module to rout calls. Anyway freeswitch was crashed two times I have attached core dump on “ https://pastebin.freeswitch.org/view/4463025f ” Our freeswitch version is “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)” Please advise to solve this matter asap. -- Best Regards.. Achintha Udukumbura

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postgres db contains non existent channels
by Roman Dissauer 01 Apr '18

01 Apr '18

Hi, we have about 800 connected users, 150 simultaneous calls and some user agents send sip messages with non UTF-8 characters. When FreeSWITCH tries to alter the channel information in the postgres db it throughs an error: 2018-03-29 15:47:46.794346 [ERR] switch_pgsql.c:656 Error executing query: ERROR: invalid byte sequence for encoding "UTF8": 0xf6 0x63 0x68 0x6c The Problem now is that we see some calls in the database which do not exist anymore (uuid_exists returns false). This are other calls, not the one which cause the error above! It seems that everytime the error happens, Freeswitch cannot delete other channels from db when the bye is arriving +/- 100ms near the error. We tried to set client_encoding in the postgres connection string to some options e.g. LATIN1 but that didn’t solve our problem. Whichever client_encoding we used, we always had some user agents with some unknown characters. How should we deal with that? Thanks, Roman RDI SOLUTIONS e.U. Hollenthon 105 2812 Hollenthon, Österreich T: +43 1 3530349 - 10 F: +43 1 3530349 - 99 roman.dissauer(a)rdi.at www.rdi.at

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javascript calculate duration of call
by Francesco Facco de Lagarda 31 Mar '18

31 Mar '18

Good morning all, I will be using both digital (which knows when the other part answers) and analogue (when, at least in Italy, doesn't!) I need to calculate the total duration of an outgoing call. Calls also must have a preset max duration. I need to schedule a warning message (1 min before timeout) and a hangup (on timeout) for BOTH digital and analogue lines. This is the core of the code, reduced to the bone: var allowedSecs = 300; // max length 5 min . session.answer(); var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060"); if (trunkIsDigital) { sessOut.execute("set", "execute_on_answer_1=sched_hangup " + allowedSects + " alloted_timeout") sessOut.execute("set", "execute_on_answer_2=sched_broadcast + " + (allowedSects-10) + " playback::" + soundDir + "one_min_left.wav both"); } else { // will schedule message and hangup on allowedSecs + graceTime to be calculated according to destintation (time to ring/answer) } if (sessOut.ready()) { bridge(session, sessOut); } sessOut.hangup(); session.hangup(); /*** HOW LONG DID THE CALL (bridge) ACTUALLY LAST?? .. for billing purposes **/ Thanks and Happy Easter!

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freeswitch segfault with mod_python
by Do Nguyen Ha 31 Mar '18

31 Mar '18

hi list i am testing freeswitch with mod_python and freeswitch sometime dead :( Mar 30 23:05:08 localhost kernel: freeswitch[1961]: segfault at 0 ip 00007f97a936ec31 sp 00007f97a4b24ba8 error 4 in libc-2.17.so [7f97a92e8000+1b8000] [root@localhost ~]# rpm -qi glibc.x86_64 Name : glibc Version : 2.17 Release : 196.el7_4.2 Architecture: x86_64 Install Date: Sat 03 Feb 2018 09:48:58 PM EST Group : System Environment/Libraries Size : 14107624 License : LGPLv2+ and LGPLv2+ with exceptions and GPLv2+ Signature : RSA/SHA256, Sat 02 Dec 2017 09:35:03 AM EST, Key ID 24c6a8a7f4a80eb5 Source RPM : glibc-2.17-196.el7_4.2.src.rpm Build Date : Thu 30 Nov 2017 01:53:33 PM EST Build Host : c1bm.rdu2.centos.org Relocations : (not relocatable) Packager : CentOS BuildSystem <http://bugs.centos.org> Vendor : CentOS URL : http://www.gnu.org/software/glibc/ Summary : The GNU libc libraries Name : freeswitch Version : 1.6.20 Release : 1.el7.centos Architecture: x86_64 Install Date: Sun 11 Mar 2018 04:34:58 AM EDT Group : Productivity/Telephony/Servers Size : 12776118 License : MPL1.1 Signature : DSA/SHA1, Wed 24 Jan 2018 02:16:20 PM EST, Key ID d76edc7725e010cf Source RPM : freeswitch-1.6.20-1.el7.centos.src.rpm Build Date : Wed 24 Jan 2018 02:14:46 PM EST Build Host : b-centos7-2.freeswitch.org Relocations : /usr Packager : Ken Rice Vendor : http://www.freeswitch.org/ URL : http://www.freeswitch.org/ Summary : FreeSWITCH open source telephony platform

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need help to run python script with mod_python
by Do Nguyen Ha 31 Mar '18

31 Mar '18

hi i user python_example.py script to test with freeswitch https://gitlab.k-net.fr/shimaore/freeswitch/blob/c6be531b4b6c38b69274057b19… i installed freeswitch-python and check the scripts folder is in : [root@localhost site-packages]# fs_cli -x 'global_getvar'|grep script script_dir=/usr/share/freeswitch/scripts and the freeswitch-python package: [root@localhost site-packages]# rpm -ql freeswitch-python /etc/freeswitch/autoload_configs /etc/freeswitch/autoload_configs/python.conf.xml /usr/lib/python2.7/site-packages/freeswitch.py /usr/lib/python2.7/site-packages/freeswitch.pyc /usr/lib/python2.7/site-packages/freeswitch.pyo /usr/lib64/freeswitch/mod/mod_python.so /usr/lib64/python2.7/site-packages/ESL.py /usr/lib64/python2.7/site-packages/ESL.pyc /usr/lib64/python2.7/site-packages/ESL.pyo /usr/lib64/python2.7/site-packages/_ESL.so where do i place the python-example.py script for freeswitch working properly Thank you

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Raw SIP request text in Lua (or any other) script
by Eugene Prokopiev 30 Mar '18

30 Mar '18

Hi, Is it possible to get raw SIP request (INVITE or so) text with headers and body as is in Lua (or any other) script? -- WBR, Eugene Prokopiev

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Setting hold_music variable
by Peter Baines (lists) 30 Mar '18

30 Mar '18

Hello, I am trying to set hold music per channel but cannot override the default that is set in vars.xml. I am setting the default hold_music variable in vars.xml, then in a dialplan I am overriding like so: <extension name="prefs_2007" continue="true"> <condition field="${sip_to_uri}" expression="2007(a)pbaines.com"> <action application="log" data="WARNING I have matched preferences for user [2007(a)pbaines.com]"/> <action application="set" data="hold_music=/var/lib/mymusic.wav"/> <action application="export" data="hold_music=/var/lib/mymusic.wav"/> <action application="bridge_export" data="hold_music=/var/lib/ mymusic.wav"/> </condition> </extension> I can see it hitting the logline in fs_cli, I am them putting it on hold like so: <extension name="soft_hold"> <condition field="destination_number" expression="^soft_hold$"> <action application="log" data="WARNING putting on hold with ${hold_music}" /> <action application="set" data="hold_music=${hold_music}" /> <action application="export" data="hold_music=${hold_music}" /> <action application="soft_hold" data="4"/> </condition> </extension> The music from vars.xml is played, even though the log line above tells me that ${hold_music} is set to my custom wav. I have also tried the following which also plays the default music on hold: <action application="soft_hold" data="4 silence ${hold_music}"/> Should I be able to overwrite the hold_music variable ? Regards, Peter

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List update... gmail addresses with plus signs are banned.
by Brian West 29 Mar '18

29 Mar '18

If you have subscribed via a gmail.com address to the mailing list in the format of user+tag(a)gmail.com, You're ok, but I had to block all gmail addresses that contain a plus sign, it was being abused, 1000's of emails were being added and used to target specific gmail users so their email would stop working because we were flooding them with subscribe confirmation emails. It was a total of three target addresses coming from 1000's of different IP addresses. This was also adding to our bounce issue with google. Thanks, -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwg…>

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Re: [Freeswitch-users] Chat between SIP and Verto
by Matthew Grooms 28 Mar '18

28 Mar '18

Hello everyone, Does anyone have any insight into how 'SIMPLE MESSAGE' delivery can be achieved between sip and verto clients? The verto documentation is very light on detail especially when it comes to messaging. There are references to using chat in conferences but that's about it. I've tried appending the protocol in the "to" value ( ie. "verto+9905(a)voip.gh.bf" ) but that doesn't appear to help much. It does change the to_proto to verto, but the message doesn't end up being delivered to the verto client. Any help would be greatly appreciated. -Matthew On 3/21/2018 3:03 PM, Matthew Grooms wrote: > > Hello All, > > We have a few hundred agents on a custom built call center platform > that leverages freeswitch and mod_callcenter. Up until now, agents > have used a custom built Windows Client based on pjsip. I'm trying to > replace that with a new client built with WebRTC and mod_verto. The > last piece of the puzzle I'm trying to solve is sending chat messages > between Verto and SIP clients. I can use the chat command from fs_cli > to send messages to freeswitch to a Verto client, but I can't seem to > send messages from a SIP client to a Verto client or a Verto client to > a SIP client. After spending a few days on this, I thought I'd appeal > to the list to see if someone can help provide some guidance. > > # show registrations > reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata > > 1000,voip.gh.bf,0d5831a1b9274063a1e06237f47a68af,sofia/external/sip:1000@X.X.X.X:49524;transport=TCP;ob,1521650573,X.X.X.X,49524,tcp,voip2.gh.bf, > > > # verto status > Name Type > Data State > ================================================================================================= > > mine profile ws:0.0.0.0:8081 RUNNING > mine profile wss:0.0.0.0:8082 RUNNING > mine::9905@voip.gh.bf client X.X.X.X:60992 CONN_REG (WSS) > ================================================================================================= > > 1 profile , 1 client > > > Messages are delivered when I try the following ... > > CHAT TO VERTO: > -------------------- CONSOLE -------------------- > # chat verto|1000@voip.gh.bf|9905@voip.gh.bf|test > 2018-03-21 11:20:48.170147 [ALERT] mod_verto.c:604 WRITE X.X.X.X:49548 [{ > "jsonrpc": "2.0", > "id": 6, > "method": "verto.info", > "params": { > "msg": { > "from": "1000(a)voip.gh.bf", > "to": "9905(a)voip.gh.bf", > "body": "test" > } > } > }] > -------------------- CHROME -------------------- > onVertoMessage from:1000@voip.gh.bf to:9905@voip.gh.bf body:test > > > Messages are not delivered when trying to send from client to client ... > > SIP TO VERTO: > -------------------- EVENT -------------------- > KV -> Chat-Send-From: "A" > <sip:1000@voip.gh.bf>;tag=759225c356d545d2b8c83b284063cac2 > KV -> Chat-Send-Profile: external > KV -> Chat-Send-To: 9905(a)voip.gh.bf > KV -> Content-Length: 4 > KV -> Core-UUID: 74a1329b-6a85-4078-9c0d-f9af5972bf0a > KV -> Error-Reason: no recipient > KV -> Error-Type: chat > KV -> Event-Calling-File: sofia_presence.c > KV -> Event-Calling-Function: sofia_presence_chat_send > KV -> Event-Calling-Line-Number: 276 > KV -> Event-Date-GMT: Wed, 21 Mar 2018 15:44:13 GMT > KV -> Event-Date-Local: 2018-03-21 10:44:13 > KV -> Event-Date-Timestamp: 1521647053570639 > KV -> Event-Name: CUSTOM > KV -> Event-Sequence: 299577 > KV -> Event-Subclass: sofia::error > KV -> FreeSWITCH-Hostname: voip2.gh.bf > KV -> FreeSWITCH-IPv4: X.X.X.X > KV -> FreeSWITCH-IPv6: ::1 > KV -> FreeSWITCH-Switchname: voip2.gh.bf > -------------------- CONSOLE -------------------- > 2018-03-21 10:44:13.570639 [WARNING] sofia_presence.c:221 Not sending > to local box for 9905(a)voip.gh.bf > 2018-03-21 10:44:13.570639 [ERR] sofia_presence.c:272 Chat proto [sip] > from ["A" <sip:1000@voip.gh.bf>;tag=759225c356d545d2b8c83b284063cac2] > to [9905(a)voip.gh.bf] > test > Nobody to send to: Profile external > > VERTO TO SIP: > -------------------- EVENT -------------------- > KV -> Content-Length: 22 > KV -> Core-UUID: 74a1329b-6a85-4078-9c0d-f9af5972bf0a > KV -> Event-Calling-File: mod_verto.c > KV -> Event-Calling-Function: verto__info_func > KV -> Event-Calling-Line-Number: 3374 > KV -> Event-Date-GMT: Wed, 21 Mar 2018 15:50:56 GMT > KV -> Event-Date-Local: 2018-03-21 10:50:56 > KV -> Event-Date-Timestamp: 1521647456550625 > KV -> Event-Name: MESSAGE > KV -> Event-Sequence: 299780 > KV -> FreeSWITCH-Hostname: voip2.gh.bf > KV -> FreeSWITCH-IPv4: X.X.X.X > KV -> FreeSWITCH-IPv6: ::1 > KV -> FreeSWITCH-Switchname: voip2.gh.bf > KV -> Nonblocking-Delivery: true > KV -> dest_proto: GLOBAL > KV -> from: 9905(a)voip.gh.bf > KV -> from_full: 9905 > KV -> from_host: voip.gh.bf > KV -> from_user: 9905 > KV -> proto: verto > KV -> to: 1000(a)voip.gh.bf > KV -> type: text/plain > KV -> verto_jsock_uuid: 85eef9c8-54c2-f4a7-eb95-ef9f5f1e6ff8 > KV -> verto_profile: mine > -------------------- CONSOLE -------------------- > freeswitch(a)voip2.gh.bf> 2018-03-21 10:50:56.550625 [ALERT] > mod_verto.c:1384 READ X.X.X.X:64275 [{ > "jsonrpc": "2.0", > "method": "verto.info", > "params": { > "msg": { > "to": "1000(a)voip.gh.bf", > "from": "9905(a)voip.gh.bf", > "body": "This is a test message" > }, > "sessid": "85eef9c8-54c2-f4a7-eb95-ef9f5f1e6ff8" > }, > "id": 5 > }] > 2018-03-21 10:50:56.550625 [ALERT] mod_verto.c:604 WRITE X.X.X.X:64275 [{ > "jsonrpc": "2.0", > "id": 5, > "result": { > "message": "SENT", > "sessid": "85eef9c8-54c2-f4a7-eb95-ef9f5f1e6ff8" > } > }] > > Thanks in advance, > > -Matthew >

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using ssh connection in module python in dialplan
by Morteza Bashsiz 27 Mar '18

27 Mar '18

Hi everybody I have an asterisk(172.16.1.33) and freeswitch(172.16.1.30) My scenario : 1. Calls come to freeswitch ivr menu 2. In ivr calls transfer to asterisk after running an python script freeswitch start recording voice 3. In asterisk calls push to an queue my python script is : ############################################################ import paramiko hostname = '172.16.1.33' port = 3022 username = 'root' pkey_file = '/root/.ssh/id_rsa' def get_agent_id(num): command="asterisk -rx \"pjsip show channels\" | grep -a1 \"CLCID: \\\"%s\\\"\" | grep Channel | cut -d \/ -f2 | cut -d\- -f1 " % (num) key = paramiko.RSAKey.from_private_key_file(pkey_file) s = paramiko.SSHClient() s.load_system_host_keys() s.connect(hostname, port, pkey=key) stdin, stdout, stderr = s.exec_command(command) result=str(stdout.read()) s.close() return result #if __name__ == "__main__": # agent_id=get_agent_id('1000') # print (agent_id) import freeswitch as freeswitch def handler(session, args): caller_num = session.getVariable("caller") agent_num = get_agent_id(caller_num) session.setVariable("AGENT_NUM", agent_num); session.setVariable("CALLER_NUM", caller_num); ############################################################ my dial plan is : ############################################################ <extension name="test_dialplan"> <condition field="destination_number" expression="^2605$"> <action application="playback" data="phrase:shatel_before_queue"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="caller=${caller_id_number}"/> <action application="python" data="test"/> <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}_${AGENT_NUM}.wav"/> <action application="bridge" data="sofia/internal/261910@ $${asterisk_vip}"/> </condition> </extension> ############################################################ *MY PROBLEM IS:* when i want to use my variable AGENT_NUM it is null BUT when i use the python manually it returns true value please help me what is my mistake ??????

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