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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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ASR dialplan.xml example for mod_unimrcp
by Chris Morrow 23 Dec '14

23 Dec '14

Hello, I am trying make a simple freeswitch->unimrcp speech recognition test with dialplan.xml. I have been experimenting recently with freeswitch and mod_unimrcp. I have checked the documentation on website: - https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp and search extensively through the unimrcp and freeswitch lists, and google, but without success. I have successfully installed and tested: - freeswitch [with direct recording via dialplan] - mod_unimrcp - unimrcp [with test unimrcpserver and umc client] There is examples of implementing a dialplan.xml with a tts_engine, and some javascript or other examples for the asr, but I cannot seem to correctly figure out how to make a full, end-to-end example of the dialplan.xml based asr unimrcp call. Ideally I like to just replicate the simple test in unimrcp, but via freeswitch cal [basically the 'run recog']l: - http://www.unimrcp.org/manuals/html/InstallationManual.html#_Toc391758895 I think I can probably figure out the rest if that works. Thanks for your feedback! Chris

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Fax tone detection
by Ali Pey 23 Dec '14

23 Dec '14

Hello, Is there anyway to make fax tone detection more sensitive? I use spandsp_start_fax_detect for the first 10 seconds of the call and fax tones are not always detected. It seems that if the tone is not loaud enought, freeswitch doesnt' detect it. Thanks, Ali Pey

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Bleg transfer
by David Villasmil Govea 23 Dec '14

23 Dec '14

Hello Guys, I receive a call from side A and send it out to side B. I need to unbridge this call once it is answered and send the B side to a queue to give it moh. How do I do that? I'm doing everything with lua and I've tried: on my dialplan I have: <action application="set" data="execute_on_answer=lua /usr/local/freeswitch/scripts/check_answered.lua"/> in check_answered-lua, after checking the call was in fact answered I do: session:execute("transfer", "-bleg 9999 XML default"); and on my dialplan I have: <extension name="9999"> <action application="log" data="INFO SENDING CALL TO QUEUE >>>>>>>>>>>>>>>>>>"/> <action application="lua" data="/usr/local/freeswitch/scripts/queue.lua"/> <action application="hangup" data="NORMAL_CLEARING"/> </extension> queue.lua: session:execute("sched_hangup","+50 alloted_timeout"); session:execute("callcenter","agents_queue"); session:execute("sleep",my_dur); But this doesn't seem to work at all... Can anyone give me a hand? thanks! David

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Freeswitch video one to many
by Thomas Auge 22 Dec '14

22 Dec '14

Hello list :-) does Freeswitch offer functionality to distribute streams to multiple listeners/viewers without conferencing (one way)? Thanks, Thomas

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Bookmarking Audio recording
by Moishe Grunstein 22 Dec '14

22 Dec '14

I am working on creating a hotline for playback of children stories. One of their requirements are bookmarks, if a child wants to stop in middle of a story and call back the next day, they should be able to resume playback from the location they stopped. I see Freeswitch has the ability to set the start position using @@, so I can resume playback from 3 minutes in to the story for example https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+playback I am not clear if there is any method to see the position that the caller is currently at, so I can save his position to a DB when he bookmarks it. I was thinking of using the call duration however that is complicated to calculate as I am planning to use seek & speed options https://wiki.freeswitch.org/wiki/Session_streamFile. Does anyone know of a method to capture the place in a wav or mp3 file currently playing? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

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strange issue in video call
by Stefano Sasso 22 Dec '14

22 Dec '14

Hi guys, I am encountering a strange issue in video calls. Please note that I cannot use proxy-media or bypass-media, because I need to have audio transcoding. (I tried with version 1.4.14 - but encountered the same problem in 1.2.x) So, in my sofia profile configuration I have as inbound-codec-prefs and outbound-codec-prefs the following string: OPUS,PCMU,PCMA,GSM,H264 I have the late-codec-negotiation and renegotiate-codec-on-reinvite set to true. I am NOT using early audio. (just to clarify) when A starts a call in audio only mode (only m=audio in sdp), on the B-leg I always see also the m=video sdp part. So, after the call is established, if from A I start sending a video, the B UAC receives it. But, if I start sending the video from B, there is no re-invite (I think it's because the B UAC thinks the video channel is already up), so the video ends in freeswitch (A UAC never receives the video - it has only a m=audio channel!). If I start the call with the video from the beginning, everything works fine. I also tried to remove the H264 from the outbound-codec-prefs, but in that case the reinvite with the video does not work. Is there a way to fix that behaviour? (send the m=video in the B-leg only if present in the A-leg, and then accept the re-INVITEs with the m=video from A-leg or B-leg) thanks! bests, stefano -- Stefano Sasso http://stefano.dscnet.org/

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ping
by Abdul Hakeem 22 Dec '14

22 Dec '14

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Lua Retry Call if call Fail
by Massimo Varriale 22 Dec '14

22 Dec '14

Hi, I'm trying to use Lua to create a callflow for my call. In particular, I would like to try a call to another gateway if the first fail. I found this wiki (https://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause) but doesn't worked for me. What could be the right procedure? Thank you Max

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