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Developing with FreeSWITCH - getting started
by rick.jarvis＠magicmail.mooo.com 14 Oct '15

14 Oct '15

I want to create an app for call handling on FreeSWITCH, in particular stacking calls, connecting to audio channels on a sound card in response to mouse clicks etc. What’s the best way to get started? In particular: What are my options, API / Language wise? I’m familiar with Python, Bash, and Basic. Are these options? Any tips, or pointers to useful ‘idiots guides’ / ‘howtos’ …? Thanks! RICK

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Freeswitch ignoring BYE sent by LegA when ignore_early_media=True
by Sushil Tripathi 14 Oct '15

14 Oct '15

I need to play a hold music file on LegA while Freeswitch tries to bridge call to one of the listed numbers. Here in dial plan I have used only one number. If I use following dialplan without ignore_early_media=true in bridge, Leg A is able to disconnect the call when Freeswitch is trying to bridge the call to one of the number in legB, but if ignore_early_media is true in that case BYE sent by Leg A is ignored by freeswitch causing call to keep ringing at LegB. <extension name="SIP-TO-SIP"> <condition field="destination_number" expression="^91(\d+)$"> <action application="answer"/> <action application="set" data="hold_music=/srv/sounds/123.mp3"/> <action application="bridge" data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/> <action application="hangup"/> </condition> </extension> While debugging for the root cause of this issue we found that this bug is because of side-effect of commit-id "3e00d270d7477a63b3c2e845b5473e6e7e3f1cfc" which was added to fix bug https://freeswitch.org/jira/browse/FS-6006 . Now we are not sure whether to remove this patch or not. Can you please suggest how to proceed further. Thanks & Regards, Sushil Tripathi +91-9910700668 -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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No SIP UPDATE will send
by Alexander Haugg 14 Oct '15

14 Oct '15

Hi all, I have the problem, that the freeSwitch don't send a SIP UPDATE to the PBX. Scenario: PBX (Siemens H3K) is connected via SIP trunking to the FreeSwitch. If I have two Legs from the H3K to the Freeswitch I'm using the uuid_bridge command to connect this two legs (works fine), but the call legs will not updated via SIP UPDATE. The PBX have no information to update the displays on the telephones. The target number (toHeader) was the routing number from the H3K to the freeswitch. Now I need a SIP UPDATE from FreeSwitch to PBX to actualize the toHeader with the new partner number (the partner number is the fromHeader of the other leg). In the sofia profile I was try it with the parameter <param name="send-display-update" value="true"/> and without this parameter (in the source code the default is "true") What can I do to force an UPDATE to the call legs? Version of the FreeSwitch FreeSWITCH Version 1.5.15b+git~20150121T080154Z~90ab1d16f5~64bit (git 90ab1d1 2015-01-21 08:01:54Z 64bit) In the INVITE from the PBX the ALLOW header includes the UPDATE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE Many thanks!

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Re: [Freeswitch-users] Help with compiling libyuv 32bits
by Luis Daniel Lucio Quiroz 13 Oct '15

13 Oct '15

BTW, has anybody been able compile it in 32 bits? Le 2015-10-13 12:28 PM, "Mario" <mario_fs(a)mgtech.com> a écrit : > Thank you, a question: How does this differ from the one in the support > lib? The URL will go into the wiki and possibly a homebrew formula. > > I was able to make install it into /usr/local instead of /usr but > freeswitch still does not find it. Will look into it further. > > > On Oct 12, 2015, at 10:48 AM, Michael Jerris <mike(a)jerris.com> wrote: > > 509 git clone https://mikej@freeswitch.org/stash/scm/sd/libyuv.git > 510 cd libyuv > 512 make > 514 sudo make install > > just did on my machine and it worked fine? > > On Oct 12, 2015, at 1:39 PM, Mario <mario_fs(a)mgtech.com> wrote: > > I got nowhere this weekend trying to compile libyuv on OS X so will wait > for Michael’s magic. I am also looking into making the new prereqs homebrew > formulas and want to see if it’s possible for libyuv. If so, I was thinking > of doing a homebrew formula called “freeswitch-support/prereqs” with > everything in it. > > > On Oct 8, 2015, at 6:31 PM, Michael Jerris <mike(a)jerris.com> wrote: > > what does this have to do with this thread? > > On Thursday, October 8, 2015, Thomas Quintana <thomas(a)bettervoice.com> > wrote: > >> https://hub.docker.com/r/bettervoice/freeswitch-container/ >> >> Thomas Quintana >> Chief Technology Officer >> Phone: 1 (512) 677-6200 >> Website: http://www.bettervoice.com >> >> >> >> On Thu, Oct 8, 2015 at 3:54 PM, Mario <mario_fs(a)mgtech.com> wrote: >> >>> Thanks I look forward to it since I am closer to 1.6! >>> Mario G >>> >>> > On Oct 8, 2015, at 12:50 PM, Michael Jerris <mike(a)jerris.com> wrote: >>> > >>> > You don't need the utilities, I'm going to have to go through and >>> figure out what all i needed to do manually and how I did it and write it >>> up for you... I'll try to do this soon. >>> > >>> >> On Oct 8, 2015, at 2:44 PM, Mario <mario_fs(a)mgtech.com> wrote: >>> >> >>> >> I git cloned libyuv from the FreeSWITCH repos into OS X. The official >>> libyuv google doc states I need the google chromium utilities. But >>> searching the web I see FreeSWITCH folks are just using “make”. There is no >>> configure file but make seems to work. But make install fails since it does >>> not know the target directories. The question is: >>> >> >>> >> Do I need the chromium utilities to build the FreeSWITCH libyuv >>> version or can I just use make/make install and how to specify targets. I >>> know it’s OS X (BSD) but Linux should be close enough. Thanks, >>> >> Mario G >>> >> >>> _________________________________________________________________________ >>> >> 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-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 >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > 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-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 >>> >>> >>> _________________________________________________________________________ >>> 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-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 >>> >> >> _________________________________________________________________________ > 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-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 > > > _________________________________________________________________________ > 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-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 > > > _________________________________________________________________________ > 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-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 > > > > _________________________________________________________________________ > 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-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 >

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How to originate a SIP call without SDP
by Gustavo Garcia 13 Oct '15

13 Oct '15

Is it possible to originate a SIP call from Freeswitch without SDP in the INVITE? (SDP offer/answer will happen in the 200 / ACK) I have tried to play with the enable-3pcc variable without luck. Any help is very welcomed. Thx a lot, G.

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Gateway State
by Leonardo Lima Ribeiro 13 Oct '15

13 Oct '15

Hello all! I need to know how many calls are running for each gateway (in my case outbound calls). When I do sofia status I have the running number of calls for the external profile like RUNNING (1), but at the gateway line of the result, the state remains REGED. I tried too sofia status gateway, but no success. Here’s my Sofia status: freeswitch@internal> sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::example.com gateway sip:joeuser@example.com NOREG 192.168.10.72 alias internal ALIASED external profile sip:mod_sofia@192.168.10.72:5080 RUNNING (1) external::example.com gateway sip:joeuser@example.com NOREG external::example.com gateway sip:joeuser@example.com NOREG external::5_JIVE gateway sip:**********@domain.jive.rtcfront.net REGED external::4_NEXTIVA gateway sip:*******@sbc.voipdnsservers.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.10.72:5060 RUNNING (1) ================================================================================================= 4 profiles 1 alias And my sofia status gateway, here I can’t see even the number of calls that is running: freeswitch@internal> sofia status gateway Profile::Gateway-Name Data State Ping Time IB Calls(F/T) OB Calls(F/T) ================================================================================================= external-ipv6::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::5_JIVE sip:**********@domain.jive.rtcfront.net REGED 0.00 0/0 0/0 external::4_NEXTIVA sip:*****@sbc.voipdnsservers.com REGED 0.00 0/0 0/2 ================================================================================================= 5 gateways: Inbound(Failed/Total): 0/0,Outbound(Failed/Total):0/2 So, I just want to know if my current call is using 5_JIVE or 4_NEXTIVA. In this actual scenario I can know that my calls are being made by Nextiva because I have 0/2 outbound calls there and for Jive I have 0/0… But if I have any number for Jive, I will not know anymore... Any idea? Thank you, Leonardo Ribeiro

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FreeSWITCH Week in Review (Master Branch) October 3rd-October 10th
by Ken Rice 12 Oct '15

12 Oct '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1jZsp7A FreeSWITCH Week in Review (Master Branch) October 3rd-October 10th Hello, again. This past week in the FreeSWITCH master branch we had 51 commits! There were some very important changes this week to the Debian packaging system. The default is now set to build packages with the upstream FS package repos. Since the system dependencies have been removed from the FS codebase the 1.6 branch is now required to use the FS public repo for dependencies. The notable feature for this week is the addition of the variable media_mix_inbound_outbound_codecs, which mixes inbound and outbound codecs, and this is a behavior change. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8290 [verto_communicator] Automatically mark dedicated encoder if out/in bandwith isn’t set to ‘Server default’ FS-8290 [verto_communicator] Adding help text on how to enable dedicated remote encoder FS-8321 [core] Add variable media_mix_inbound_outbound_codecs to mix inbound and outbound codecs. BEHAVIOR CHANGE Improvements in build system, cross platform support, and packaging: FS-8316 [build][Debian] Fixed new build warning from latest clang and resolved the build warnings in the modules too FS-8271 [Debian] Adding some logging, and more cautious handling of spaces in parameters. Now the default will build packages with the upstream FS package repos. This is a change in the default behavior of the Debian packaging system with the justification that in 1.6 it is now required to use the FS public repo for dependencies because system dependencies have been removed from the FS codebase which used to be included. And defaulting to automatically download the binary dependencies because without major changes to package building in cowbuilder(which is the primary supported method of building FS packages), you can’t access the network to build the binary packages from the source package. If using system apt repo list, then include the supplementary ones too FS-8233 [automation] In order to clean up build dependencies for the automated tests, convert the tests/*/Makefile.am into an include file for the top level Makefile.am. This will greatly simplify dependency tracking, and allow tests to be rerun easily on FS source code changes. FS-7820 [automation] Use a more appropriate function for printing diagnostics The following bugs were squashed: FS-8243 [mod_opus] Adding back the missing part removed in 8b088c2 so FEC works in most surroundings FS-8295 [mod_opus] FMTP fixes to continue the cleanup of FEC FS-8302 [mod_opus] Fix some printing/logging because switch_opus_show_audio_bandwidth() was not returning TRUE/FALSE as expected FS-8130 FS-8305 [mod_opus] Fix some warnings and errors caused by dtx and/or jittery webrtc, refactor of last patch, and add suppression of scary harmless message about opus FEC FS-8296 [mod_opus] Improve the way Opus is initialized when a call comes in FS-8179 [mod_opus] Fixed a regression setting fec_decode breaking output on stereo calls FS-8297 [mod_conference] A fix for auto STUN switching IPs quickly and WebRTC video not working FS-8130 [mod_conference] Fix for micro cut-offs and unstable voice issues FS-8317 [mod_conference] Fix for playing multiple files at once to stack them for immediate playback, sometimes breaking and the floor layer becoming unusable for the rest of the conference. FS-8067 [verto_communicator] When no email is present make sure mm is the default avatar in the circle this way the talk indicator works on PSTN and SIP callers. FS-8247 [verto_communicator] When websocket disconnects go to splash screen to wait for the reconnect FS-8300 [verto_communicator] Fixing reload bug so reloading twice is no longer needed FS-8315 [core] Fix for rtp_media_timeout not working FS-8304 [core] Fix for choppy audio during calls FS-8320 [core] Fixed broken ZRTP not responding to HELLO packet FS-8311 [mod_voicemail] Fix for leave-message event not containing verbose data for a forwarded voicemail FS-8318 [mod_av] Fix for recording being out of sync when video from chrome has packet loss FS-7929 [mod_sofia] Fixed an issue when processing SIP messages while using camp-on FS-6833 [mod_sofia] Add content-type header to ack with sdp And, this past week in the FreeSWITCH 1.4 branch we had 3 new commits merged in from master. And the FreeSWITCH 1.4.23 release is here! Go check it out! The following bugs were squashed: FS-8246 [mod_json_cdr] Use seconds as default value for delay parameter FS-8282 [core] Fix for sleep is not allowing interruption by uuid_transfer FS-8166 [core] Mute/unmute while shout is playing audio fails because the channel “has a media bug, hard mute not allowed”

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FreeSWITCH in the RTC Quick Start Guide
by Daniel Pocock 12 Oct '15

12 Oct '15

FreeSWITCH is mentioned in a few places, including the PBX chapter, in the RTC Quick Start Guide[1] I've put up a blog today about my work on this book and some questions[2] for discussion. I'd be particularly interested in any feedback from the FreeSWITCH community about just how FreeSWITCH fits into the federated SIP and RTC environment and whether this book makes it easier for people. Contributions are welcome too, either through Github or feel free to email me. Regards, Daniel 1. http://rtcquickstart.org 2. http://danielpocock.com/rtc-quick-start-becoming-a-book-now-in-beta

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add yourself to planet.sip5060.net
by Daniel Pocock 12 Oct '15

12 Oct '15

Hi all, If you are blogging about open source and SIP, I'll be happy to syndicate it on http://planet.sip5060.net - this is good for everybody doing open source SIP, because it helps things go up in Google and it makes it easy for people to follow the whole community in one place. All syndication links back to your blog Please just send me an email with the config for me to cut and paste, the RSS feed URL in brackets and then the name to appear next to your blogs, here are some examples: [http://danielpocock.com/rss.xml] name = Daniel Pocock [http://www.kamailio.org/w/feed/] name = Kamailio I already added a few people, they are listed at the bottom of the planet page. If only some of your blogs are about SIP, please consider tagging the SIP blogs and giving me a feed URL for the tag. Regards, Daniel

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Freeswitch inbound logic match
by Gonzalo Gasca Meza 12 Oct '15

12 Oct '15

I have a SIP Provider which has 4 IP addresses in US. They send SIP calls from any of those 4 IPs to my Freeswitch. **They do not require authentication, nor SIP Trunk registration, just purely send a SIP INVITE** Incoming calls *work fine* except that I want to assign a *context* when I receive an incoming call from this ITSP at *gateway level*. Gateway xml file is configured under external folder. (../conf/sip_profiles/external/) I can see in packet capture and in freeswitch.log call comes from correct ip and port, but is always routed to context default. Hence I need to configure something there. Is it possible to define context at gateway level? Console trace: http://pastebin.com/NzzLAK8U Freeswitch trace http://pastebin.com/YUYVLfyY I defined my 4 SIP Gateways (status up) as follows: (1 for each IP address) <gateway name="itsp-inbound-us1"> <param name="auth-calls" value="false"/> <param name="proxy" value="54.172.60.0"/> <param name="register" value="false"/> <param name="context" value="itsp"/> <param name="username" value="not-required"/> <param name="password" value="not-required"/> <param name="from-user" value="not-required"/> <param name="expire-seconds" value="600"/> <param name="extension" value="1000"/> </gateway> >sofia profile external gwlist itsp-inbound-us4 itsp-inbound-us3 itsp-inbound-us2 itsp-inbound-us1 > sofia status gateway itsp-inbound-us1 Name itsp-inbound-us1 Profile external Scheme Digest Realm 54.172.60.0 Username not-required Password yes >From <sip:not-required@54.172.60.0> Contact <sip:gw+itsp-inbound-us1@52.2.15.172:5060 ;transport=udp;gw=twilio-inbound-us1> Exten 1000 To sip:not-required@54.172.60.0 Proxy sip:54.172.60.0 Context itsp Expires 600 Freq 600 Ping 0 PingFreq 0 PingTime 0.00 PingState 0/0/0 State NOREG Status UP Uptime 536s CallsIN 0 CallsOUT 0 FailedCallsIN 0 FailedCallsOUT 0 ================================================================================================= Any suggestion?

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