October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call Recording in freeswitch
by bhavik patel 11 Oct '14

11 Oct '14

Hi, I want to record each and every call in some specific format of file. I already tried with record_session application but no luck. Can any one suggest me how can i do this using FS ? -- Thanks, Bhavik Patel

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Re: [Freeswitch-users] Callee name before connect
by Chris B. Ware 11 Oct '14

11 Oct '14

Brian, setting <action application="set_profile_var" data="callee_id_name=Alice"/> fixed in part my problem. Now in 180 Ringing I have Remote-Party-ID: "Alice" <sip:201@mydomain.com>;party=calling;privacy=off;screen=no It makes Polycom phones happy, but Snom want pai, not rpid. Is there a way to change? Thank you Date: Tue, 7 Oct 2014 12:55:16 +0100 From: Andrew Cassidy <andrew(a)cassidywebservices.co.uk> Subject: Re: [Freeswitch-users] Callee name before connect To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Message-ID: <CAMbW9oKhnm9+cHxA3tmA-ep_=S+qyOB=mcj0KwkkBm9HCBnicA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" No, bridge should handle the session states from there. The point of using ring_ready is it forces freeswitch to send an early 180 - Ringing response where bridge would normally copy the 180 or 183 response from the b-leg. It's worth mentioning the other thing I do related to caller id stuff is use the extension number as the username, makes sure it gets set by the devices properly. On 6 October 2014 22:20, Blake Priddy <bpriddy(a)bryantschools.org> wrote: > Andrew would you do that after the bridge as well? > > On Mon, Oct 6, 2014 at 11:32 AM, Andrew Cassidy < > andrew(a)cassidywebservices.co.uk> wrote: > >> I find <action application="ring_ready" /> can also help >> >> On 6 October 2014 15:59, Brian West <brian(a)freeswitch.org> wrote: >> >>> just call set_profile_var callee_id_name before the bridge. >>> >>> On Mon, Oct 6, 2014 at 9:35 AM, Chris B. Ware <chrisbware(a)yahoo.it> >>> wrote: >>> >>>> Hi, >>>> >>>> I'm using FS 1.2.24. My customer wants the following feaure: >>>> >>>> - Bob is 201 >>>> - Alice is 202 >>>> - When Bob calls 202, wants to see "Alice" on his phone's display on >>>> Ringing >>>> >>>> It works after connect but not during provisional response. According >>>> to Wiki I'm using >>>> >>>> <action application="bridge" >>>> data="{origination_callee_id_name=Alice}user/202@${domain_name}"/> >>>> >>>> Any help? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian(a)freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F >> <info(a)cassidywebservices.co.uk> *03300 100 961 >> *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk >> *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F <info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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[Announcement] FreeSWITCH 1.4.12 Released!!
by Ken Rice 10 Oct '14

10 Oct '14

Hi FreeSWITCHers! The FreeSWITCH Team would like to announce FreeSWITCH 1.4.12! This is a bug fix release and addresses a possible DoS. Thanks to Travis Cross for finding and patching this issue. commit b5294c53d694499fe0f4a807554154ca8c14dfa6 Author: Travis Cross <tc(a)traviscross.com> Date: Fri Oct 10 18:25:11 2014 +0000 Fix crash on transport=tls with non-TLS profile We use the transport of the Contact header of the remote UAC to decide which of our own Contact addresses we should use when replying to a SUBSCRIBE or sending a presence NOTIFY. If TLS is not enabled on a Sofia profile, then the TLS Contacts for that profile are NULL. Unfortunately we were using these NULL values uncritically when the remote UAC sent us a Contact header with a TLS transport and our own Sofia profile did not have TLS enabled. With this commit we fall back to our TCP Contact address when the remote Contact is TLS and our Sofia profile does not have TLS enabled. There are a few other minor bug fixes also to address a media resume after hold problem that is limited in scope and fixes to fs_encode for G722 file encoding. Source tarballs are available via http://files.freeswitch.org/ and packages are available via the FreeSWITCH Yum and Debian repositories. Have a Great Weekend! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Can't get port audio to work
by Paradise aka Mr Glasspoole 10 Oct '14

10 Oct '14

Hi, after almost 20 years internet this is the first time I'm using a mailing list so please bear with me if I'm doing something wrong. Is there a way to format stuff (code, quotes)? I'm trying to use FreeSwitch as headless phone on a Banana Pi. I have Bananian running (pure Debian) and used the Raspberry Raspbian APT repository. I guess it should work without problems cause its both Wheezy. There where no errors on the installation... I also have FreeSwitch Debian netinstall running on my old Thinkpad that i use for testing. After trying 2 hours to get port audio running on the Banana Pi i thought i try it first on the FreeSwitch Debian netinstall. But i have the same problems. "pa devlist" just says: -ERR no reply What i added to the netinstall is: aptitude install alsa alsa-base alsa-utils alsa-tools alsaplayer-alsa libasound2-dev alsa-oss mpg321 And i did "modprobe snd_pcm_oss" and "modprobe snd_mixer_oss" and freeswitch is also member of audio. Sound is working and before i saw you can use FreeSwitch as softphone i was playing with pjsip and there python modul and that works. But the question is what is less demanding on the Banana Pi and thats the reason i want to try it. Regards from Germany Björn

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Unable to find mod-ilbc on debian repository
by Douglas Lopes Pereira 10 Oct '14

10 Oct '14

Hi everyone, I've being using debian repository ( http://files.freeswitch.org/repo/deb/debian/) to install/update freeswitch on my ubuntu box. For some reason I can no longer find mod-ilbc in there. I could not find any reason for that on the mailing list (neither on internet) and the module page (https://wiki.freeswitch.org/wiki/Mod_ilbc) does not mention anything about deprecation. Does anyone knows why this package is no longer provided? Regards, Douglas

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is callee/caller id update possible if a call is already answered?
by Hristo Trendev 10 Oct '14

10 Oct '14

Hi, Does FS support caller and callee id updates when a call leg is already answered? The following test cases didn't work for me: == Case 1 == A sends an invite to FS and is bridge to B as follows (pseudo dial plan code): answer <-- this is important sleep 500ms bridge(B) B sends a P-Asserted-Identity header in the 180 Ringing, which is detected by FS because it prints in the CLI: "Update Callee ID to "B" <123>" However, no sip message is sent on the A-leg, which I suspect is due to the fact that the A-leg is already answered. == Case 2 == I use originate to call A and when A answers the call, I bridge it to B: originate sofia/internal/A(a)1.1.1.1 &bridge(sofia/internal/B(a)2.2.2.2) Again B send a PAI header in the 180 Ringing and again FS prints a "Update Callee ID to "B" <123>" line in the log, however nothing is ever sent to A. == Case 3 == 1. A is connected to a conference (call is active) 2. During the call, A sends a re-invite with P-Asserted-Identity header, which contains the updated caller id information. 3. This doesn't seem to be supported at all, because the CID remains unchanged and nothing interesting is printed in the CLI. My question is: are these cases (or maybe some of them) actually supported/implemented or am I doing something wrong? I am testing with a vanilla freeswitch configuration, no changes whatsoever. Regards, Hristo

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 10 Oct '14

10 Oct '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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valet_park slot occupation status
by Stanislav Sinyagin 10 Oct '14

10 Oct '14

hi, it would be great to have a function which would return true or false if a valet park slot is occupied. Then, in the pickup extension, we would play a sound indicating that the parking lot is free, instead of placing the call into the parking lot. Should I file a JIRA ticket for this enhancement?

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[ANNOUNCEMENT] FreeSWITCH 1.4.10 Released!
by Ken Rice 10 Oct '14

10 Oct '14

The FreeSWITCH Team is happy to announce the release of 1.4.10 Also released is are sound sets 1.0.51! Source is available at: http://files.freeswitch.org/freeswitch-1.4.10.tar.bz2 Or via stash.freeswitch.org for Git users. Updates include nearly 70 patches addressing 20+ Jira Tickets, 9 Pull Requests. As well as the addition of new features such as ability to filter the SDP on bypass_media calls to remove or limit codecs. Sounds updates include updated normalization of volume levels, minor adjustments to volume levels, and fixing of a possible bug in the last packaging of the Music On Hold Files. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Voicemail Behavior
by Siri MM 09 Oct '14

09 Oct '14

Hi, I am currently using Freeswitch 1.2.7. I have setup a couple of extensions, with voicemail enabled as follows: Under directory, extension xml has vm-enabled set to true Under dialplan, I have the following: <extension name="901" > <condition field="context" expression="public" > </condition> <condition field="destination_number" expression="^901$" > <action application="ring_ready" /> <action application="set" data="call_timeout=10" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="bridge" data="user/901" /> <action application="answer" /> <action application="voicemail" data="default $${domain} 901" /> </condition> </extension> Now, lets say I dial the above extension 901 from another extension, say 902. After 10 seconds, the call is redirected to voicemail: a. Voice prompt says: "Person at extension 901 is not available. Record message after the tone. Press any key or stop talking to end the recording" b. At this stage, if the caller (902) records a message, and presses any key other than '*', then the normal voicemail submenu continues (i.e. Press 1 to listen, 2 to save etc..) c. However, if the caller (902) were to record a message and press '*', instead of the above voicemail sub menu, he actually gain access to the callee's Voicemail box!! Caller 902 is able to listen to 901's messages, change settings etc. Could anyone please let me know if I am missing something here?

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