January 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to get sip_to_tag as session variable
by John Nash 07 Jan '16

07 Jan '16

I am trying to get sip_to_tag session variable in a lua script which is executed on media (183) I can see from tag call id but not to tag I saw this was fixed in https://freeswitch.org/jira/browse/FS-7137 but I am using higher version FreeSWITCH Version 1.4.26+git~20151124T001158Z~45534616c6~64bit (git 4553461 2015-11-24 00:11:58Z 64bit)

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Digest-realm for hidden caller-id
by Volodymyr Fedorov 07 Jan '16

07 Jan '16

Hi community, I experience interesting issue. Freeswitch running in multi-domain environment. UA generates invite with anonymous(a)anonymous.invalid in FROM which is quite normal according to rfc3325. Sample bellow: === INVITE sip:44264@pbx-sumy.co.ia.com SIP/2.0 Via: SIP/2.0/UDP 10.10.6.170:5062;branch=z9hG4bK1491276326;rport From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1989469488 To: <sip:44264@pbx-sumy.co.ia.com> Call-ID: 1409580995-5062-3(a)BA.BA.G.BHA CSeq: 20 INVITE Contact: "Anonymous" <sip:54240@10.10.6.170:5062> Max-Forwards: 70 User-Agent: Grandstream GXP1625 1.0.3.6 Privacy: id P-Preferred-Identity: <sip:54240@pbx-sumy.co.ia.com> Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 332 === Freeswitch sends 407 and uses as Digest-realm domain part from FROM : "Proxy-Authenticate: Digest realm="anonymous.invalid"" Example: === SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.6.170:5062;branch=z9hG4bK1491276326;rport=5062 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1989469488 To: <sip:44264@pbx-sumy.co.ia.com>;tag=ZamrKr7UecpNp Call-ID: 1409580995-5062-3(a)BA.BA.G.BHA CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26-37~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.win , message-summary, refer Proxy-Authenticate: Digest realm="anonymous.invalid", nonce="917b30c9-9070-4ae5-87e6-1814b379fd94", algorithm=MD5, qop="auth" Content-Length: 0 === And call fails 403 Forbidden. So main question how to rid this. And is it feature or bug. Maybe I can use domain part from P-Preferred-Identity header ? Thank you in advance. -- Best regards, Volodymyr

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Re: [Freeswitch-users] Make of FreeSWITCH 1.6 on CentOS 7 issue(s) - resolved
by Andrew Keil 07 Jan '16

07 Jan '16

To all, Found what I was missing (and causing all my build issues): rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm Now when I perform the yum install commands and it pulls down the right version and everything works as expected. Here is my procedure (for those who wish to use the manual configure/make approach): Install FreeSWITCH 1.6.0 on CentOS 7 (fresh installation): Note: Everything below is run inside CentOS 7 terminal as "root" user (via "su -" command) 1) Make sure yum update is run and CentOS is up-to-date 2) {optional} inside Terminal window select Edit Profile Preferences and select "Use custom default terminal size" and make it 160 x 60. Close terminal and re-open for the change to take affect. 3) yum -y install git 4) yum -y install epel-release 5) rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm 6) yum -y install git gcc-c++ autoconf automake libtool wget python ncurses-devel zlib-devel libjpeg-devel openssl-devel e2fsprogs-devel sqlite-devel libcurl-devel pcre-devel speex-devel ldns-devel libedit-devel libxml2-devel 7) yum -y install libyuv-devel opus-devel libvpx-devel libvpx2* libdb4* libidn-devel unbound-devel libuuid-devel lua-devel libsndfile-devel 8) cd /usr/local/src 9) git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git 10) cd /usr/local/src/freeswitch 11) ./bootstrap.sh 12) ./configure 13) make 14) make install 15) make cd-sounds-install 16) make cd-moh-install 17) cd /usr/local/freeswitch/conf 18) vi vars.xml Change: <X-PRE-PROCESS cmd="set" data="default_password=1234"/> {!!set it to something different!!} Save and close (<Esc> :wq!) 19) cd /usr/local/freeswitch/conf/sip_profiles 20) mv internal-ipv6.xml internal-ipv6.xml.removed {disables ipv6 support} 21) mv external-ipv6.xml external-ipv6.xml.removed {disables ipv6 support} 22) cd /usr/local/freeswitch/bin 23) ./freeswitch -nonat -nonatmap 24) ... to exit from FreeSWITCH console FreeSWITCH Version: 1.6.0+git~20150903T203652Z~6762f14140~64bit (git 6762f14 2015-09-03 20:36:52Z 64bit) From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Andrew Keil Sent: Saturday, 19 September 2015 12:06 PM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Make of FreeSWITCH 1.6 on CentOS 7 issue(s) This worked! However mod_opus fails now. Since I cannot perform a yum install libopus-devel do I need to follow this link: http://files.freeswitch.org/downloads/libs/opus-1.1.tar.gz Getting there! Andrew From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Andrew Keil Sent: Saturday, 19 September 2015 8:46 AM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] Make of FreeSWITCH 1.6 on CentOS 7 issue(s) Thanks I will give this a try. On 19 Sep 2015, at 12:07 AM, Christopher Rienzo <cmrienzo(a)gmail.com<mailto:cmrienzo@gmail.com>> wrote: sudo yum install lua-devel On Fri, Sep 18, 2015 at 1:39 AM, Andrew Keil <andrew.keil(a)visytel.com<mailto:andrew.keil@visytel.com>> wrote: To FreeSWITCH users, I am attempting to make the latest FreeSWITCH release on CentOS 7 (I am trying to avoid the RPM approach since I am also working with Sangoma to assist with their updated merge of freetdm into FreeSWITCH). CentOS 7: CentOS-7-x86_64-DVD-1503-01.iso (with yum update done successfully). As root user: yum install git gcc-c++ autoconf automake libtool wget python ncurses-devel zlib-devel libjpeg-devel openssl-devel e2fsprogs-devel sqlite-devel libcurl-devel pcre-devel speex-devel ldns-devel libedit-devel libxml2-devel {also did the following to see if these will help} yum install libyuv-devel yum install libvpx-devel yum install libidn-devel yum install unbound-devel yum install libdb4* yum install libuuid-devel cd /usr/local/src git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git cd freeswitch ./bootstrap.sh vi modules.conf {comment out mod_fsv (due to libyuv-devel error)} ./configure ./make making all mod_lua make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/languages/mod_lua' CXX mod_lua_la-mod_lua.lo mod_lua.cpp:37:17: fatal error: lua.h: No such file or directory #include "lua.h" The mod_fsv issue I do not care so much about however Lua I need. FYI: I did try the rpm approach on a separate CentOS 7 VM and noticed the Lua dependant package is version 5.1.4 not 5.2. I hope someone has an idea of what to try next. Regards, Andrew Keil _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@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<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org

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switch_r_sdp change for 183 and 200 OK
by John Nash 06 Jan '16

06 Jan '16

I am trying to experiment few things that required me to change SDP (Using media bypass mode). I am able to change switch_r_sdp from a /perl/lua script successfully for Invite message using ... $session->execute("set","switch_r_sdp=$$resp{sdp}"); but I also want to change SDP of 183 session progress message I tried executing same code in scripts fired on 183 message but it doesnt seem to be changing SDP Is that even possible? John

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Re: [Freeswitch-users] lua IVR interactivity problem
by Ray Keating 06 Jan '16

06 Jan '16

> I get a 403 error when trying to access your page. Thanks for the feedback. The mod_managed/C# PizzaDemo direct download page URL should have been: http://www.pontimax.com/#/downloads (Dang, SPA "#"s) You can of course just go to www.pontimax.com and select "Downloads" -Ray

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Freeswitch "Cluster"
by Pierre-Emmanuel Dutang 06 Jan '16

06 Jan '16

Hello, I need to create a cluster of Freeswitch for a project. the project is simple, we call people through freeswitch and we want to bridge 2 sessions, pretty simple :) But to be more scalable I want to use multiple freeswitch, that mean a call can go through one freeswitch F1 and another call through freeswitch F2. What I want is to be able to bridge a call from F1 with a call from F2. The solution can be bridge both sessions from 2 different freeswitch or move a channel to be in the same freeswitch to be able to bridge both sessions. I have no idea if I can do that and how to do it. I just read some stuff about mods_cluster but nothing clear with explanation. Thanks for you help :) -- Pierre 06 52 08 00 52 <http://www.linkedin.com/in/pedutang>

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How to get the answered call live duration?
by ankitdoshi 06 Jan '16

06 Jan '16

Hello, How can I get the answered call live duration ? Let me give an example what i want to ask, let's say user A dialed a call to B and B receives it after 10 seconds, they talked for 20 seconds so now i am getting 30 seconds in cdr, i want it exact 20 seconds Now let me explain what I did so far that I am getting this o/p I am doing current time - created_epoch but its giving total duration of ringing + answered duration (as per exp, 30 secs) Let me provide you the 'show calls' info while call ringing, uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch bce09d00-6**1-4**7-baa1-803f633b57cd,inbound,2016-01-06 19:30:26,14520***26,sofia/default/48******81@*.*.*.*,CS_EXECUTE,48******81,48******81,*.*.*.*,12345678,48******81@*.*.*.*,,RINGING,,,,,server,,,,,,,,,,,,,,,,,,,,, 97e47575-5**4-4**6-91cb-b6e75fea713e,outbound,2016-01-06 19:30:26,14520***26,sofia/default/12345678,CS_CONSUME_MEDIA,48******81,48******81,*.*.*.*,12345678,,,RINGING,Outbound Call,12345678,,bce09d00-6**1-4**7-baa1-803f633b57cd,server,,,,,,,,,,,,,,,,,,,,, And also 'show detailed_bridged_calls' info while call is on going uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_application,b_application_data,b_dialplan,b_context,b_read_codec,b_read_rate,b_read_bit_rate,b_write_codec,b_write_rate,b_write_bit_rate,b_secure,b_hostname,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_call_uuid,b_sent_callee_name,b_sent_callee_num,call_created_epoch bce09d00-6**1-4**7-baa1-803f633b57cd,inbound,2016-01-06 19:30:26,14520***26,sofia/default/48******81@*.*.*.*,CS_EXECUTE,48******81,48******81,*.*.*.*,12345678,,,XML,default,PCMA,8000,64000,PCMA,8000,64000,,servertwo,48******81@*.*.*.*,,HELD,,,,bce09d00-6**1-4**7-baa1-803f633b57cd,,,97e47575-5**4-4**6-91cb-b6e75fea713e,outbound,2016-01-06 19:30:26,14520***26,sofia/default/12345678,CS_EXCHANGE_MEDIA,48******81,48******81,*.*.*.*,12345678,playback,local_stream://moh,XML,default,PCMA,8000,64000,PCMA,8000,64000,,servertwo,,,ACTIVE,Outbound Call,12345678,,bce09d00-6**1-4**7-baa1-803f633b57cd,,,1452088836 Here, everywhere i am getting same time '2016-01-06 19:30:26' as created_epoch so is there any variable like 'answered_epoch' or 'bridged_epoch' which provide the exact time of answered call duration (as per exp, 20 secs) ? and not provide the ringing + answered duration (as per exp, 30 secs)? Any insights would be helpful. Thanks. ----- Thanks Ankit Doshi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-get-the-answered-call-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] lua IVR interactivity problem
by Ray Keating 06 Jan '16

06 Jan '16

>> Does anyone use a scripting language based IVR and have this solved? >> Or otherwise have a solution that doesn't involve maintaining 2 >> distinct IVRs If I correctly understand your requirements, then, yes there is a solution. Dave Kompel of Xarix.net has a set of C# classes that wrap the mod_managed FreeSWITCH/FreeSWITCH.Native Event and FreeSWITCH core interface routines to provide the capabilities that you are seeking. His ivr.cs class includes a Play() method that has overloads for playing a prompt file or a list of prompt files. A GetDTMF() method fully supports DTMF barge-in, as well as integrated speech recognition for mixed speech recognition/DTMF based barge-in. Helper classes for dial plan and directory configuration and handling are provided as is FreeSWITCH event accessing and processing. He makes the source freely available for redistribution. We have implemented a mod_managed, C# based version of the classic "Pizza Ordering" speech recognition demo, using Dave's C# classes and our recently introduced MRCPv1, Microsoft Speech Platform 11 based speech recognition server, "mrcpSP11-STT". It fully illustrates the use of single prompt/multiple prompts, prompt barge-in and integrated speech and DTMF handling. It also provides multiple examples of the use of his recently added ASR interfacing capability, via session exec of "detect_speech" and ASR event tracking. A Visual Studio solution containing Dave's C# classes and of course the IVR application classes is available for download at www.pontimax.com/downloads. The IVR prompt files are not contained in the application solution but can be provided upon request. Ray Keating www.pontimax.com 203-526-0170 On Mon, Dec 28, 2015 at 5:41 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote: > it's much easier to implement the IVR logic via an ESL connection. > Then you can script it in your preferred programming language and > perform all necessary database lookups or whatever your application > requires. > > But in both cases (Lua or mod_perl or mod_python or an ESL > application), you would need to execute playAndGetDigits in order to > let the user enter DTMF during the playback of the prompt. If you have > multiple audio files to play back, you just execute playAndGetDigits > with every audio file in the sequence until the DTMF input is > detected. You can also build a phrase macro and use it instead. > > In case of ESL, you can also subscribe to DTMF events and play audio > files asynchronously if playAndGetDigits is not sufficient for your > needs. > > Here is a short example in Golang, probably you'll find it useful: > https://github.com/xlab1/go-fs-secretary-prototype/blob/master/golang/secre… > > > > > On Mon, Dec 28, 2015 at 6:46 PM, Peter Hartmann > <peter at hartmanncomputer.com> wrote: >> I'm trying to port an IVR to lua. I'm new to lua by the way. The big >> picture: the goal is that the IVR should change based on the output of >> a shell script. >> >> I've followed the basic example over here, substituting my several wav files >> for the TTS prompts. >> >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+Welcome+IVR+example >> >> But I've noticed a huge difference between this and the core IVR >> functionality. DTMF will not interrupt the prompts, it will only be >> accepted when the script reaches the session:getDigits line. It >> seems like session:playAndGetDigits could be interactive, but it only >> accepts one wav. Is there a way for either playAndGetDigits to accept >> multiple wav files (without concatinating)? Or otherwise is there a >> way for a scripted IVR to have this interactive functionality? Seems >> like I need to fork the getDigits first and then play the wavs, right? >> >> I realize I could probably do the logic in the dialplan and then send >> the caller to different IVRs, but I'd rather not maintain 2 mostly >> similar IVRs. >> >> Does anyone use a scripting language based IVR and have this solved? >> Or otherwise have a solution that doesn't involve maintaining 2 >> distinct IVRs? >> >> Thanks much >> >> >> >> >> >> Peter Hartmann >> Hartmann Computer Consulting >> http://blog.hartmanncomputer.com >> (212)203-8870 >> >> If I can't explain it to you in plain language, that means I don't >> understand it. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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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digit_action_set_realm question
by Abaci B 06 Jan '16

06 Jan '16

Hi, I was wondering if there is any way to unset the bind_digit_action realm once it's set using digit_action_set_realm, I tried setting it to "_UNDEF_" and "" none which seem to work.

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Authentication of REGISTERs through a RADIUS server
by Daniel Corbe 05 Jan '16

05 Jan '16

Hello, I took note of mod_xml_radius but the documentation on this module is thin in comparison to the rest of the mod set. It isn’t clear to me how I’d tell FreeSWITCH to begin routing REGISTER requests to my RADIUS server once I’ve enabled and configured the mod. I’ve got a pretty simple configuration in place as I’m only interested in registrations (below). Is there something I need to do now either from the dialplan or the Sofia config? Someone hitting me with a cluebat here would be lovely. <configuration name="xml_radius.conf" description="Radius XML Gateway"> <auth_reg> <connection name="testing"> <param name="authserver" value="192.168.164.137:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/local/src/freeswitch/src/mod/xml_int/mod_xml_radius/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="request-type=user"/> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param name="User-Name" variable="sip_from_user" format="%s"/> <param name="Digest-Response" variable="sip_auth_response" format="%s"/> <param name="Digest-Realm" variable="sip_auth_realm" format="%s"/> <param name="Digest-Nonce" variable="sip_auth_nonce" format="%s"/> <param name="Digest-Username" variable="sip_auth_username" format="%s"/> <param name="Digest-URI" variable="sip_auth_uri" format="%s"/> <param name="Digest-Method" variable="sip_auth_method" format="%s"/> <param name="Digest-Algorithm" variable="sip_auth_method" format="MD5"/> <param name="Digest-Qop" variable="sip_auth_qop" format="%s"/> <param name="Digest-CNonce" variable="sip_auth_cnonce" format="%s"/> <param name="Digest-Nonce-Count" variable="sip_auth_nc" format="%s"/> </fields> </auth_reg> <configuration> freeswitch(a)voice-lab.hammerfiber.net> load mod_xml_radius 2016-01-05 12:43:33.179332 [INFO] mod_xml_radius.c:170 Could not find 'auth_invite' section in config file. 2016-01-05 12:43:33.198334 [INFO] mod_enum.c:880 ENUM Reloaded 2016-01-05 12:43:33.198334 [INFO] switch_time.c:1415 Timezone reloaded 1781 definitions 2016-01-05 12:43:33.198334 [INFO] mod_xml_radius.c:257 Could not find 'auth_app' section in config file. 2016-01-05 12:43:33.198334 [INFO] mod_xml_radius.c:291 Could not find 'acct_start' section in config file. 2016-01-05 12:43:33.198334 [INFO] mod_xml_radius.c:325 Could not find 'acct_end' section in config file. 2016-01-05 12:43:33.198334 [CONSOLE] switch_loadable_module.c:1538 Successfully Loaded [mod_xml_radius] 2016-01-05 12:43:33.198334 [NOTICE] switch_loadable_module.c:292 Adding Application 'radius_auth' 2016-01-05 12:43:33.198334 [NOTICE] switch_loadable_module.c:338 Adding API Function ‘xml_radius_debug'

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