December 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Zoiper registration on freeswitch
by Mujahid Ali 24 Dec '14

24 Dec '14

hi all, i install freeswitch via yum from http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_I… and zoiper on my system (CentOS 6) but when i tried to register Zoiper to freeswitch with defaults user like username: 1001 password: 1001 and click on register i got this on freeswitch cli *freeswitch@internal> 2014-12-24 05:41:18.421648 [CONSOLE] mod_voicemail.c:4066 Event Thread Started* and on zoiper *SIP 403 - Forbidden.* but zoiper not register even i disable SElinux and got same thing. how can i register zoiper with my freeswitch ? thanks in advance. NOTE: all installation is on my local PC freeswitch and zoiper etc.

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Receives call From Unknown Extensions
by Lloyd Aloysius 24 Dec '14

24 Dec '14

Hi All I have a multi domain setup. We receive calls from unknown extensions (eg: 100 , 101,1000,1007 etc ).But there is no voice in it. We do not have any default extensions in the system and all default extensions removed from the system. Users are authenticated by alphanumeric (like an email username) Eg: mike@ mydomain.com and passwords are very complicated. How someone can call a user without authentication from these extensions? Please let me know how to solve this issue. Thanks Lloyd

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[Mod CDR CSV] Adding Timestamp into the CDR
by Massimo Varriale 24 Dec '14

24 Dec '14

Hi Guys, I would like to write a field into the CSV with the timestamp of the CDR so I can know the writetime. The timestamp should be in this format "2014-12-22 10:32". Is this possible? I digged the wiki but I didn't found any clue regarding this. Thank you Max

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Differrent codec parameters (fmtp) per leg
by Thomas Auge 24 Dec '14

24 Dec '14

Is it a supported configuration to have different formats for the same codec on each end? I.e. for Opus if one leg sends stereo and the other doesn't, or if they send with different bitrates? I built a webRTC app a while ago, which works like that, and it's not a problem in Chrome. But as I started to work with Freeswitch I realized this is probably not the way it is meant to be done. :-) Would it cause issue or do some codecs not care what they are being fed for decoding?

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[Outbound call using GSMOpen] help needed for research on Maternal Health
by Abdul Mannan Butt 23 Dec '14

23 Dec '14

Hi Freeswitch-Users, We are basically working on a research project on Maternal Health, in which we have to call to women and instruct them about the diet at each stage of their pregnancy to reduce mortality rate due to pregnancy complications. Could anyone please help us in implementing an outbound call? We are trying the following code but we are unable to make an outbound call properly and play a sound file when call get received by the called person. *Code:* -------------------------------------------------------------------------------- audio_file_path = "ivr/MESSAGE_TO_PLAY.wav";freeswitch.consoleLog("INFO","Initiating call\n"); *-- *this will call the person and move to next line of call get accepted/rejected etc *session = freeswitch.Session("{ignore_early_media=true}gsmopen/gsm01/03454329512");* *-- *Checking session status session:consoleLog("info", "checking session status\n"); -- Stream audio if session is ready if(session:ready() == true) then session:consoleLog("info", "ready to stream\n"); session:streamFile(audio_file_path); session:sleep(250); session:hangup();end -------------------------------------------------------------------------------- But the problem is *session = freeswitch.Session("{ignore_early_media=true}gsmopen/gsm01/03454329512"); *is not returning the control to the below code until user disconnect the call. At that time session got invalid. please help, how could we do Call-Out properly. We have posted the following issue n Jira but we remain unable to get our work done. https://freeswitch.org/jira/browse/FS-7101 https://freeswitch.org/jira/browse/FS-7057 Please help us to make an outbound call and play a audio message. We will be very thankful to you. Kind Regards, Amna Batool and Abdul Mannan, Information Technology University, Lahore Pakistan

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[ERR][mod_json_cdr.c][Got error [500] posting to web server]
by Bilal Abbasi 23 Dec '14

23 Dec '14

Hi All, I am using Mod_json to store cdr in MySQLDB, i have successfully configured that and data is successfully inserted in Database. But i got this error on CLI. I don't know why i get this error and how can i get ride of that. *2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:396 Got error [500] posting to web server [http://localhost/cgi-bin/bilal.py <http://localhost/cgi-bin/bilal.py>]* *2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:403 Retry will be with url [http://localhost/cgi-bin/bilal.py <http://localhost/cgi-bin/bilal.py>]* *2014-12-23 10:03:53.157185 [ERR] mod_json_cdr.c:414 Unable to post to web server* Regards Bilal Abbasi

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Fwd: webrtc chrome->sip bad media jssip error
by rentmycoder rentmycoder 23 Dec '14

23 Dec '14

I've checked with 1.4 latest git, it works using latest SIPMl5 and SipJs in all directions. But now FireFox latest (34.05) is broken with both js stacks... Browser->SIP is working, but SIP->Browser and Browser->Browser is broken... So FS 1.4.14 does not work with latest Chrome, 1.4.master does not work with latest FF. Maybe SDP issue... On Thu, Dec 18, 2014 at 3:22 PM, rentmycoder rentmycoder < rentmycoder(a)gmail.com> wrote: > Hi, > > Maybe it's a know issue, please direct me to the right direction. > Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web > sip client, > > Issue: > Using Firefox, calls are working great, but with Chrome 39.0.2171.95m > calling from browser to SIPendpint fails. > The sip client rings, but after answer, the jssip client shows bad media > description error.... > > Ring SDP: > v=0 > o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 > E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host > generation 0 > > > 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=1001 8000 8000 IN IP4 192.168.101.39 > s=SIP Call > c=IN IP4 192.168.101.39 > t=0 0 > m=audio 5004 RTP/AVP 0 13 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > Remote SDP: > 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP > sofia/internal/1002(a)192.168.101.43: > v=0 > o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 > s=FreeSWITCH > c=IN IP4 192.168.101.43 > t=0 0 > a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > m=audio 27200 RTP/SAVPF 0 126 106 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 > E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 > a=rtcp-mux > a=rtcp:27200 IN IP4 192.168.101.43 > a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k > a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 > a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ > a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 > a=ice-ufrag:lO4wBcSvCchx5t6B > a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT > a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host > generation 0 > >

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Mute without conference
by Paul Atreides 23 Dec '14

23 Dec '14

How can I implement music on hold without using a conference? I want to turn off the microphone so that the leg B can not hear what leg A is saying. Does anyone know? Thank you

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Openldap and freeswitch integration problem
by Shisheer Teli 23 Dec '14

23 Dec '14

Hi, I am able to bind with any alise on ldap server except userPassword (MD5) alise. when i bind password with userPassword , authentication fails. I done some following testing Test 1: when i set openldap userPassword in md5 , in freeswitch cli i saw hash password and authentication failed. Test 2: when i set openldap userPassword in plain text, in freeswitch cli i can see plain text password and authentication success. Authentication works with plain text but not for encrypted password. Configuration file: <configuration name="xml_ldap.conf"> <bindings> <binding name="directory"> <param name="basedn" value="dc=example,dc=com"/> <param name="filter" value="(telephoneNumber=%d)" bindings="directory"/> <param name="url" value="ldap://ldap.example.com"/> <param name="binddn" value="cn=use,dc=example,dc=com"/> <param name="bindpass" value="XXXX"/> <trans> <tran name="id" mapfrom="uid"/> <tran name="password" mapfrom="userPassword"/> </trans> </binding> </bindings> </configuration> Please reply ASAP... -- Regards, Shisheer T

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lua xml-handler-binding and chatplan
by Luis Daniel Lucio Quiroz 23 Dec '14

23 Dec '14

Hello Just wondering if this is possible to let a custom xml-handler, written in lua to deal with chatplan as it does for dialplan. Any comment? i was reading about hoo event tag, maybe with that? LD

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