January 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SPEECH_DETECT EVENT
by Raimundo Pérez Nieves 18 Jan '18

18 Jan '18

Hi, Does speech_detect event in outbound event socket detect when a person answer or answer machine receive a call? Thanks

1 0

Re: [Freeswitch-users] Originate loops Dialplan
by Raimundo Pérez Nieves 17 Jan '18

17 Jan '18

It parks the other leg, and stops running dialplan but also do not get any other events from its. There has to be a way to keep hearing from leg B in an originate avoiding loop. Thanks If you have more ideas please Enviado desde mi iPhone El 17/01/2018, a la(s) 15:25, Christoph Russow <russow(a)emtex.de> escribió: > try &park instead of 10000 at the end of your dialstring > >> Am 17.01.2018 um 15:17 schrieb Raimundo Pérez Nieves: >> Hi guys, >> Iam trying to use originate from event socket and get events from the >> number dialed in this originate. >> The problem I am having is that when I use internal phone in my server >> works great, but when I am trying to originate an outbound call to other >> server it loops in Dialplan. >> I just want to get tone detected event from that external call. >> >> my $mydialstr = '{origination_uuid=' . $uuidLegB . >> ',absolute_codec_string=G729,originate_retries=0}sofia/external/550075378355222(a)64.34.75.23 >> <mailto:originate_retries=0}sofia/external/550075378355222@64.34.75.23> >> 10000'; >> my $res = $con->api("originate",$mydialstr); >> >> The extension 10000 in non-existing, and when Dialplan end it starts >> again and again. >> Can I put another thing just for listen events for that outbound call? >> >> Thanks >> >> >> _________________________________________________________________________ >> 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 > > -- > ________________________________________________________________ > > EMTEX GmbH > Christoph Russow > Software Engineer > > Bischof-Otto-Weg 9 > D-91086 Aurachtal > Geschäftsführer: Markus Enzinger > Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal > Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 > ________________________________________________________________ > >

2 2

help
by Ali Haider 17 Jan '18

17 Jan '18

Hiiii im try where from get this package freeswitch-meta-all : Depends: freeswitch-mod-flite (= 1.9.0~628~7f9e6f3-1~xenial+1) wget -O - https://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 /freeswitch_archive_g0.pub | apt-key add - echo "deb http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/ xenial main" > /etc/apt/sources.list.d/freeswitch.list # you may want to populate /etc/freeswitch at this point. # if /etc/freeswitch does not exist, the standard vanilla configuration is deployed apt-get update && apt-get install -y freeswitch-meta-all

1 0

Originate loops Dialplan
by Raimundo Pérez Nieves 17 Jan '18

17 Jan '18

Hi guys, Iam trying to use originate from event socket and get events from the number dialed in this originate. The problem I am having is that when I use internal phone in my server works great, but when I am trying to originate an outbound call to other server it loops in Dialplan. I just want to get tone detected event from that external call. my $mydialstr = '{origination_uuid=' . $uuidLegB . ',absolute_codec_string=G729,originate_retries=0}sofia/external/550075378355222(a)64.34.75.23 10000'; my $res = $con->api("originate",$mydialstr); The extension 10000 in non-existing, and when Dialplan end it starts again and again. Can I put another thing just for listen events for that outbound call? Thanks

2 1

Re: [Freeswitch-users] Originate doubt
by Raimundo Pérez Nieves 17 Jan '18

17 Jan '18

This work sensational!!!! Thanks $mydialstr = '{origination_uuid=' . $uuid . '}sofia/external/55007537203517(a)64.34.75.23 <mailto:55007537203517@64.34.75.23> 5004'; and then $con->bgapi('originate',$mydialstr);

3 2

Attended Transfer and CallerID
by Chris B. Ware 17 Jan '18

17 Jan '18

Hi, a customer is passing from an Alcatel PBX to Freeswitch. He used to "transfer" Caller ID number together with the call during attended transfers. I mean: - A (mobile phone) calls B (freeswitch user)- B makes an attended transfer (REFER) to C (another freeswitch user)- If B hears ringback, doesn't wait C to answer and hangup (now A is ringing on C)- On C display there's B number and not A number. On Alcatel PBX C can see A number. How can I implement this feature? I've seen an old mail from Anthony suggesting to use set_profiole_var on caller_id_name but doesn't work on 1.6.19. Thank you.

1 0

ODBC and Tables
by Alexander Perkins 17 Jan '18

17 Jan '18

Hi All. I have compiled Freeswitch with ODBC support (I think) and I have also updated switch.conf.xml and external.xml, internal.xml, and internal-ipv6.xml in sip_profiles. However, when I look at my MySQL database, I do not see the auto-generated tables. When I look at the logs, I see 'SQL Enabled'- http://drops.tiltx.org/smOoiw/Cpy85pS9QR. Here are my screenshots: switch.conf.xml- http://drops.tiltx.org/n9vsT9/7fWeOLLXjh internal.xml, external.xml, and internal-ipv6.xml- http://drops.tiltx.org/WgJghY/2LDLnRps95 I am able to connect to the database vi isql. How may I verify ODBC support and what am I doing wrong? Should Freeswitch autogenerate the tables? Also, the username does have write access to that database. Thanks to All, Alex

3 2

Verto re-INVITE equivalent
by Adam Ben-Ayoun 16 Jan '18

16 Jan '18

Hi, We have mobile apps (iOS/Android) and a website that uses FreeSwitch for audio/video conferencing using mod_conference. We are currently using SIP and considering to move to Verto for signaling. In our app users are connecting without video (not sending video) that implies a=recvonly and later on they can add video dynamically using SIP re-INVITE. How can we implement this using Verto? I tried verto.attach but it doesn't really generate a new local SDP on fs.. Any idea? Thanks, Adam

2 1

WebRTC using RTP/AVP instead of RTP/SAVPF
by Alexander Haugg 16 Jan '18

16 Jan '18

Hi, in the dialplan I am using "<action application="bridge" data="{media_webrtc=true}user/123(a)my.sip.domain" />" The media description header have ever the RTP/SAVPF as set. The most Clients wich support webrtc and ICE have the possibiliti to work without SRTP (that's nice vor debugging problems). Is it possible to set RTP/AVP for webrtc calls? Thanks a lot.

2 1

(no subject)
by Ali Haider 16 Jan '18

16 Jan '18

HIii where from get this package freeswitch-meta-all : Depends: freeswitch-mod-flite (= 1.9.0~628~7f9e6f3-1~xenial+1)

2 2