March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Originate Failed
by Andre 28 Mar '13

28 Mar '13

On Windows 8, every few calls I get this error while dialing a call and then the call fails. Does anyone know what to do to fix this? [INFO] mod_dptools.c:3084 Originate Failed. Cause: NORMAL_CLEARING 2013-03-28 15:54:25.545988 [NOTICE] mod_dptools.c:3204 Hangup sofia/internal/10001(a)192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING] 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1536 Session 18 (sofia/internal/10001(a)192.168.1.148) Ended 2013-03-28 15:54:25.565992 [NOTICE] switch_core_session.c:1540 Close Channel sofia/internal/10001(a)192.168.1.148 [CS_DESTROY]

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Calling extension using ruby
by Amit Kumar 28 Mar '13

28 Mar '13

I am trying to make an application that can initiate a call from the Web app to a PSTN line. So when a user clicks on call, the PSTN line will ring, and FS will play the text entered on the web app to the PSTN. I am able to dial the PSTN from a SIP phone(Zoiper). I am also able to call the PSTN line from the CLI, but I have no clue as to how to eliminate the SIP Phone all together. I can connect to FS using ESL, and send the command to initiate a call, but since no user is registered without the SIP phone registering them, I am not sure how to go ahead. Any help is appreciated!

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mod_perl hangupCause() returning 'NONE' and hangup_cause is null
by Ashish gautam 28 Mar '13

28 Mar '13

Hi, I am running a perl script after the call lands on my dialplan. I want to get the status that whether the call was originated properly and and if yes what was the hangup cause. I am getting the hangup status using hangupCause() and it returns 'NONE' everytime no matter what the reason is. Also the hangup_cause variable is nill or not set everytime. Any help is appreciated. Thanks in advance!! -- Regards Ashish Gautam

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Access to rxfax channel variables
by Eugene Prokopiev 28 Mar '13

28 Mar '13

Hi, How to read rxfax channel variables described in http://wiki.freeswitch.org/wiki/Mod_spandsp#Controlling_the_app? My dialplan: <action application="set" data="api_hangup_hook=log INFO FAX received from ${caller_id_number}(${fax_remote_station_id}) : ${fax_result_text}"/> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/var/spool/fax/$1/${strftime(%Y%m%d%H%M%S)}-${caller_id_number}.tif"/> <action application="hangup"/> But I see only: 2013-03-28 13:28:40.256497 [INFO] mod_commands.c:5899 FAX received from 2440663() : -- Regards, Eugene Prokopiev

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ERROR: database is locked
by Steven Schoch 28 Mar '13

28 Mar '13

I just started noticing these errors: 2013-03-28 11:15:38.739806 [ERR] switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked] BEGIN EXCLUSIVE 2013-03-28 11:15:38.739806 [CRIT] switch_core_sqldb.c:1679 ERROR [database is locked] I'm a little new at sqlite. How is a database locked and how do you unlock it? -- Steve

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bin/fsapi and logging
by Cal Leeming [Simplicity Media Ltd] 28 Mar '13

28 Mar '13

Hi guys, Normally when using the freeswitch console, you are able to see all the debug/info messaging scrolling down the CLI. However, when connecting in using the fsapi binary, it does not seem to output any debugging at all. I've tried all sorts of combinations to make this work (console loglevel info / log etc), with no luck. Using "sofia loglevel all 9" makes a bunch of events related debug show, but not the info/debug messages. Any thoughts? Ca

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How to set the Job_UUID for bgapi chat command
by Leonardo Pereira 28 Mar '13

28 Mar '13

Hi. I'm using the "bgapi chat" command to send SMS: bgapi chat sip|+17777445514|external/+441797890123(a)MY_SMS_GW.COM|this is a test Is there a way to set my own Job_UUID in this case? Thanks in advance, Leo Leonardo Nogueira de Sá Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77

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FreeSWITCH on Raspberry Pi - OpenVPN gateway, B2BUA
by Peter P GMX 28 Mar '13

28 Mar '13

For those who might be interested: During the last days we built a B2BUA based on Freeswitch and Raspberry PI Ver B which connects to a central server via OpenVPN in order to encrypt voice traffic and circumvent NAT issues. We plan to use it during travelling and to have VPN access in the home office. The costs for this are relatively low (~75$). We installed * Raspbian * Freeswitch compiled native from GIT (this really takes a while) * OpenVPN * Dnsmasq for eth1 side * isc-dhcp-server for eth1 side * tftpd-hpa for eth1 side * a second LAN interface eth1 was done by a Delock USB 2.0 to LAN interface (about 20$), works out of the box * iptables firewall for NAT traversal This worked pretty well. We were not sure beforehand, whether the USB to LAN interface would work nicely, but in fact it did. CPU usage during a single call is top - 10:47:34 up 39 min, 1 user, load average: 0,64, 0,40, 0,24 Tasks: 70 total, 1 running, 69 sleeping, 0 stopped, 0 zombie %Cpu(s): 6,5 us, 12,1 sy, 0,0 ni, 64,9 id, 0,0 wa, 0,0 hi, 16,5 si, 0,0 st KiB Mem: 448776 total, 94656 used, 354120 free, 13096 buffers KiB Swap: 102396 total, 0 used, 102396 free, 42464 cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 2785 root -2 -10 28328 14m 4844 S 5,3 3,2 0:30.45 freeswitch 2651 root 20 0 5800 3052 1848 S 3,0 0,7 0:07.73 openvpn 3051 pi 20 0 5096 1256 968 R 0,7 0,3 0:01.97 top 4 GB CF card was about 75%. Codec was G711. The load was mainly generated by the USB-to-LAN adapter. Maybe someone has a better solution for this? At the end we were impressed, how easy this was to set up. So now we have an easy VPN-Router/B2BA which allows us to * connect a PC somewhere and tunnel all traffic via VPN * connect a SIP phone somewhere and tunnel all VoIP traffic via VPN We did this with an OpenWRT Box before, but this is much more easy to handle. Best regards Peter

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FS behind NAT encounter no audio
by Jimmy Chang 28 Mar '13

28 Mar '13

Hi, I could make a call within NAT without any problem. When I made a call from internet, I got no audio. I have read the document http://wiki.freeswitch.org/wiki/NAT_Traversal and set the parameters. vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.server.ip"/> internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> Any advice? Thanks. Jimmy

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Call failure with [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
by Siri MM 28 Mar '13

28 Mar '13

Hi All, I am trying a simple setup as follows: * FreeSWITCH with a SNOM phone and a X-Lite soft phone registered * Open ACL When I try to make a call from X-Lite to SNOM phone, i get a CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] error. The logs, and the configurations are at: http://pastebin.freeswitch.org/20722 Where am I going wrong?

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