April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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mod_nibblebill issue with recording
by Murugan Pandian 30 Apr '16

30 Apr '16

HI, We are using nibllebill for billing and AWS S3 for call recording storage ,When we record the call using record_session it will upload recorded WAV file to s3(AWS). Issue: 1.mod_nibblebill bill ammount include record_session wav file upload time Example: if I talk 2 mins and record_session take 20/sec to upload WAV file to S3(AWS) the billed amount 2.20/mins (0.1 per mins total amount 0.3 ) but billed time only shows 2 mins

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FreeSwitch - Clone - TAG
by Luiz Fernando Softov 29 Apr '16

29 Apr '16

Hi, I don't remember why, but I'm using 1.5.final (branch master) in production (~1 year), and it's working fine, with 30 simultaneous calls. I will re-clone the FreeSwitch, because some changes made, that i need. Someone recommend to clone another tag?

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[ANN] RoCKSwitch Builder Beta 1 released
by Richard Mace 29 Apr '16

29 Apr '16

Hi All, Just a quick note to say that our FreeSWITCH builder application is ready for further testing. It is a Windows based application, that you "point" at a freshly built Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just a few clicks. Let us know what you think http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder Richard www.rocksoftware.co.uk

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Bridge campon
by Gregor Nanger 29 Apr '16

29 Apr '16

I have strange problem with campon. Somehow FS doesn't obey what I set as campon_timeout variable. Whatever I set, it is always 60 second. It is not even default of 10 seconds. But funny is that, campon_pause works as expected. Any hints what could be wrong. I set dialplan with xml_curl, so dialplan is short and nothing is injected. Best regards, Gregor

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