- FreeSWITCH-users - FreeSWITCH Mailing Lists

Multiple inbound numbers on a registered trunk
by Joseph Waite 02 May '16

02 May '16

Hi Guys We have a Freeswitch based FusionPBX setup which registers to our main SIP Switch with Registering trunks. Outbound Calls work fine, however I’m having issues with using multiple inbound numbers. If I setup a destination with the username of the trunk then all calls received on the trunk goto that destination. If I setup each DID on its own then calls fails. In the invite I see To: <sip:trunkuser@providerip:5080> I see the DID dialled in the following in the SIP message Diversion: numberdialed <tel:numberdialed>;reason=unknown;counter=1;privacy=off Is there a setting I can add somewhere to the trunk config to tell it to use the value in Diversion? Any help greatly appreciated. Regards Joseph Waite

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SIP cause codes in CDR
by Dmitry Mordovin 02 May '16

02 May '16

Hello My VoIP provider sometimes refuse my call, for example, with 487 message (ORIGINATOR_CANCEL) or 503 (NORMAL_TEMPORARY_FAILURE) Does anyone know the way (channel variable, function call) how can I put to CDR record 487, 503 and other SIP notation cause number instead Q.850? The main reason, SIP equivalents representation simplify troubleshooting work. DM

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What are current best options for WebRTC between mobile and FreeSwitch
by Oivvio Polite 01 May '16

01 May '16

I'm working on a project that will need desktop as well as mobile clients. On the desktop side I'm using Electron[1]. On mobile I'm targeting Android and iOS but I haven't picked the development tools yet. The mobile client is a lot simpler than the desktop and on recent versions of Android I hope to get away with using Chrome. But for older versions of Android and for iOS I'll need an app. My main candidates there PhoneGap/Cordova[2] and React Native[3]. React Native seems to have a lot of promise but PhoneGap has more of a track record and there seems to be people actually using it in production for WebRTC. Since stability is really important to this project I'm leaning towards PhoneGap/Cordova. All of these tools use a Javascript runtime so I hope to be able to use verto.js for my signalling regardless of what I choose. But I'm curious about the experiences you folks have with these tools. Oivvio [1] http://electron.atom.io/ [2] https://webrtchacks.com/webrtc-hybrid-applications/ [3] https://webrtchacks.com/reacting-to-react-native-for-native-webrtc-apps-ale…

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Problem with Playback wav from remote server
by Pete Kay 01 May '16

01 May '16

Hi I am able to playback a remote mp3 but not a remote .wav file. I am getting strange error. Could someone please guide me how to resolve it? freeswitch@server3> 2016-05-01 10:37:07.751782 [DEBUG] mod_httapi.c:2631 caching: url:http://105.165.2.110/voice/m79eVzDuH6_61.wav to /usr/local/freeswitch/storage/http_file_cache/c8691a860ba4ddc0b18a39d3768add75.wav (5773 bytes) 2016-05-01 10:37:07.771763 [ERR] mod_sndfile.c:204 Error Opening File [/usr/local/freeswitch/storage/http_file_cache/c8691a860ba4ddc0b18a39d3768add75.wav] [File contains data in an unknown format.] 2016-05-01 10:37:07.771763 [ERR] mod_httapi.c:2971 Invalid cache file /usr/local/freeswitch/storage/http_file_cache/c8691a860ba4ddc0b18a39d3768add75.wav opening url 105.165.2.110/voice/m79eVzDuH6_61.wav Discarding file. 2016-05-01 10:37:07.771763 [DEBUG] sofia.c:6755 Channel sofia/internal/ 0000000000(a)105.165.2.110 entering state [early][183] Here it shows that I can "get" the .wav no problem. [root@ server3 ~]# wget http://105.165.2.110/voice/m79eVzDuH6_61.wav --2016-05-01 10:15:42-- http://105.165.2.110/voice/m79eVzDuH6_61.wav Connecting to 105.165.2.110:80... connected. HTTP request sent, awaiting response... 302 Found Location: http://105.165.2.110/auth/login [following] --2016-05-01 10:15:42-- http://105.165.2.110/auth/login Connecting to 105.165.2.110:80... connected. HTTP request sent, awaiting response... 200 OK Length: 5773 (5.6K) [text/html] Saving to: `m79eVzDuH6_61.wav' 100%[================================================================================================================================================>] 5,773 --.-K/s in 0s 2016-05-01 10:15:42 (483 MB/s) - `m79eVzDuH6_61.wav' saved [5773/5773] [root@ server3 ~]# ls -al *.wav -rw-r--r-- 1 root root 5773 May 1 10:15 m79eVzDuH6_61.wav [root@server3 ~]#

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Recording video and timing
by Luca Pradovera 01 May '16

01 May '16

Hello, after a long journey I finally managed to get my "one presenter + multiple Verto clients" setup. However, I think I might have hit some bug in the recording code. Recordings usually work well for the first 2-3 minutes, then I get a few freeze frames. After that, video is out of sync with audio and is recorded and played back at roughly 3x the correct speed. I am using the "transcode" video mode with VP8, and there seems to not be any error in the console. Where can I start debugging such an issue? Best regards, Luca Pradovera

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Freeswitch php scripts
by Nduwayezu, Joselyne 30 Apr '16

30 Apr '16

Hello, I would like to invoke php script in the dial plan. I defined the dialplan as follos: <include> <extension name="outbound-socket"> <condition field="${sip_to_user}" expression="(972550428)$"> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/> <action application="socket" data="10.0.0.6:8021 async full"/>--> </condition> </extension> </include> And my php script is: <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: playback\n"; echo "execute-app-arg: \ /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n \ "; // Wait sleep(5); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?> When i dial the DID number, the call state is "call setup" for a while, and after i'm asked to leave a message because the number is busy or unreachable. I tried different sockets in the dialplan (5060, 5080, 8040) with no results Any adeas of what is happening? Thanks NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Problem with ESL.so
by Nduwayezu, Joselyne 30 Apr '16

30 Apr '16

I,ve followed the freeswitch cookbook in order to user the ESL. I installed all the libraries needed. But when i run ESL.so, i have a segmentation fault which result in error when i run a php script. the error: unable to load dynamic library /usr/lib/php5/20131226. Any help? NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Internal profile doesn't exist
by amani mansour 30 Apr '16

30 Apr '16

Hi all, I don't know what is the problem , Remarque ,my IP address is correct in vars.xml (static) when i tape sofia status it affiche external,gateways,internal_ipv6 but the internal profile isn't exist is there someone who know the solusion please ? [image: pasted1] Best regards Amani

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"To field" in sip header
by thomas peterseil 30 Apr '16

30 Apr '16

hello, i have a sip provider with multiple DID´s and they send me all calls with the same invite, only the To field in the SIP header is different. how can i read the To field in the SIP header to be able to make a call routing based on the To field? i tried the following rule but it doesnt work: --------------------- <include> <extension name="public_test1"> <condition regex="all"> <regex field="destination_number" expression="^provider1$"/> <regex field="${sip_h_To}" expression="43717072377"/> <action application="set" data="effective_caller_id_number=7021"/> <action application="set" data="effective_caller_id_name=thomas"/> <action application="answer"/> <action application="transfer" data="6011 XML inbound1 "/> </condition> </extension> ---------------- the To field in the SIP header looks like this: To: <sip:43717072377@193.43.148.37> thanks a lot for help! best regards, thomas

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FreeSWITCH external Gateway registration Issue
by Arun K 30 Apr '16

30 Apr '16

Dear Group, I'm a novice with FreeSwitch and the below doubt is very basic. But I could not able to resolve it for the past whole week. Kindly assist me. 1. I'm trying to register my Free switch gateway with my SIP provider. the external profile is configured at C:\Program Files\FreeSWITCH\conf\sip_profiles\external\example.xml as below: 2. 3. <include> 4. <gateway name="Rcom"> 5. <param name="username" value="+914438000100"/> 6. <param name="password" value="xxxxxxx"/> 7. <param name="from-domain" value="bangalore.relianceims.in"/> 8. <param name="proxy" value="10.237.246.225"/> 9. </gateway> 10. </include> 11. 12. Registering attempts failed with below logs: 13. 14. 15. 2016-04-28 23:08:27.905875 [NOTICE] sofia_reg.c:448 Registering bangalore.relianceims.in 16. 2016-04-28 23:08:28.905875 [NOTICE] sofia_reg.c:448 Registering Rcom 17. send 596 bytes to udp/[10.237.246.225]:5060 at 17:38:28.921500: 18. ------------------------------------------------------------------------ 19. REGISTER sip:10.237.246.225;transport=udp SIP/2.0 20. Via: SIP/2.0/UDP 10.16.83.21;rport;branch=z9hG4bK11QceD1pmcUjN 21. Max-Forwards: 70 22. From: <sip:+914438000100@bangalore.relianceims.in >;tag=1H4KN7By10r8D 23. To: <sip:+914438000100@bangalore.relianceims.in> 24. Call-ID: e6089137-443f-4b24-8375-37fa7617e1c2 25. CSeq: 90596490 REGISTER 26. Contact: <sip:gw+Rcom@10.16.83.21:5060;transport=udp;gw=Rcom> 27. Expires: 3600 28. User-Agent: FreeSWITCH-mod_sofia/1.5.15b~32bit 29. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY 30. Supported: timer, path, replaces 31. Content-Length: 0 32. 33. ------------------------------------------------------------------------ 34. recv 456 bytes from udp/[10.237.246.225]:5060 at 17:38:28.937125: 35. ------------------------------------------------------------------------ 36. SIP/2.0 404 Not Found 37. Via: SIP/2.0/UDP 10.16.83.21;received=10.16.83.21;rport=5060 ;branch=z9hG4bK11QceD1pmcUjN 38. To: <sip:+914438000100@bangalore.relianceims.in >;tag=ztesipZcoVXTKT0*1-7-16648*ijib.1 39. From: <sip:+914438000100@bangalore.relianceims.in >;tag=1H4KN7By10r8D 40. Call-ID: e6089137-443f-4b24-8375-37fa7617e1c2 41. CSeq: 90596490 REGISTER 42. X-ZTE-Cause: "CSCF-1.7. BC005000-BC00DA26-BC005319-BC00521F.icscf1.bangalore.relianceims.in" 43. Content-Length: 0 44. 45. ------------------------------------------------------------------------ 46. 2016-04-28 23:08:28.937125 [ERR] sofia_reg.c:2367 Rcom Failed Registration with status Not Found [404]. failure #4 47. 48. 49. IF i try to register x-lite with SIP Provider with the following parameters it is getting registered successfully. 1. username : +914438000100 2. password : xxxxxx 3. domain : bangalore.relianceims.in 4. Authorisation user name: +914438000100(a)bangalore.relianceims.in 5. Proxy : 10.237.246.225 6. 7. Only thing i do not use at Free switch configuration that i use at X-lite is Authorisation user name. without this parameter, x-lite registeration also fails. So can any of the experts assist me as where should i provide this parameter at free switch. Else how to solve this problem. Kindly advice.

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