December 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_verto & a1-hash
by Tihomir Culjaga 19 Dec '16

19 Dec '16

i made a modification on mod_verto to allow a1-hash auth. where do i send patches ? T.

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change P-Asserted-Identity
by Andre DeMattia 19 Dec '16

19 Dec '16

HI, We have a dial plan that we require our customers to send us the Dial Number with a custom prefix, let's say 7000. We use this prefix to know where to route the call. We will strip the prefix and dial the correct number (sofia/internal/1xxxxxxxxxx;npdi=yes@xxx.xxx.xxx.xxxx:5060) On the 183 the pcap shows P-Asserted-Identity: "7000xxxxxxxxxx" <sip:+7000xxxxxxxxxx@xxx.xxx.xxx.xxx> I have no idea how that prefix got onto the P-Assereted-Identity. Can someone help me know how to change that to the real dialed number and strip the prefix?

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"FreeSWITCH" <sip:0000000000
by David Villasmil 19 Dec '16

19 Dec '16

I have in my gateway "caller-id-in-from" to true... i have in my bridge: 13 <action application="set" data="effective_caller_id_number=3051111111"/> 14 <action application="set" data="effective_caller_id_name=30511111112"/> 15 <action application="bridge" data="{origination_caller_id_number=3051111111,origination_caller_id_name=3051111111}sofia/gateway/mygw/$1"/> why do i keep seeing going out like: From: "FreeSWITCH" <sip:0000000000@1.2.3.4>;tag=Qgtcrcv3NmD4B ... Remote-Party-ID: "FreeSWITCH" <sip:0000000000@1.2.3.4 >;party=calling;screen=yes;privacy=off i seem to have forgotten or... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Handle outgoing call after answer
by Yuriy Gorlichenko 19 Dec '16

19 Dec '16

Hi. I want to make some staff (inserting in database sоme data and then transfer call) after it was answered by dialplan(does not matter is it lua dialplan or something other). I have experience with other products (pushing handlers after 200 arrived), but in freeSwitch i found only hangup_hook for handling call after it was hanged up Is it have method to do this?

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Picking up conference after second leg (called by autocall) joined
by Yuriy Gorlichenko 19 Dec '16

19 Dec '16

Hi. I have conference and for now it works with autocall to other members I want to add all members at the room only when aoutocalled members will pick up call. Is it possible by conference configuration ori will I need to create bridges by myself and then join them (redirect for example) to conference room by scripting

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Setting rpid and from for a gateway in its definition.
by Mimiko 17 Dec '16

17 Dec '16

Hello. I use sip_from_display, sip_from_uri, sip_cid_type, origination_caller_id_name, and origination_caller_id_number to dinamically rewrite from and rpid headers. Like: sip_from_display=$${default_external_exit}$${default_areacode}${effective_caller_id_number} How can I place this in gateway definition for a profile to skip specifying this in dialplan? I want to use mod_lcr and all this fields does not feet in the column. I've tried to use <param name="from-user" value="$${default_external_exit}$${default_areacode}${effective_caller_id_number}"/> But this does not change after parsing XML and ${effective_caller_id_number} is not evaluated on calling thru gateway. Also tried: <variables> <variable name="sip_from_uri" direction="outbound"><![CDATA[<sip:$${default_external_exit}$${default_areacode}${effective_caller_id_number}@$${gateway_ip_v4_bind}>]]></variable> </variables> And this does not have any effect. How can accomplish this? Thank you.

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March 23rd Cluecon Weekly lost?
by Tom Lynn 17 Dec '16

17 Dec '16

I've tried to play back the March 23rd entry of Cluecon Weekly on Youtube.com but keep getting "An error Occurred. Please try again later." Can this episode be reposted? or is it lost? Thank you. Tom

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Re: [Freeswitch-users] March 23rd Cluecon Weekly lost?
by jungle Boogie 16 Dec '16

16 Dec '16

On Dec 16, 2016 12:39 PM, "Tom Lynn" <tom(a)tomlynn.com> wrote: > > Ken, > Have you ever considered developing an Alexa Skill that could have freeswitch call someone in your google or outlook contacts? Set up a conference call amongst people, etc? > What's Alexa and what can it do with freeswitch?

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Debian 8.6 installation issue
by Murugan Pandian 16 Dec '16

16 Dec '16

Hi, When i try install freeswitch 1.6 in Debian i am getting following error in my console . The following packages have unmet dependencies: freeswitch-meta-all : Depends: freeswitch (= 1.6.13~21~e755b43-1~jessie+1) but it is not going to be installed Depends: freeswitch-meta-codecs (= 1.6.13~21~e755b43-1~jessie+1) but it is not going to be installed Depends: freeswitch-meta-mod-say (= 1.6.13~21~e755b43-1~jessie+1) but it is not going to be installed Depends: freeswitch-music Depends: freeswitch-sounds Depends: freeswitch-mod-abstraction (= 1.6.13~21~e755b43-1~jessie+1) but it is not going to be installed Depends: freeswitch-mod-avmd (= 1.6.13~21~e755b43-1~jessie+1) but it is not going to be installed Depends: freeswitch-mod-av (= 1.6.13~21~e755b43-1~jessie+1) but it is not going to be installed My Kernal Version: *3.16.0-4-amd64* My Server details: *Architecture: x86_64CPU op-mode(s): 32-bit, 64-bitByte Order: Little EndianCPU(s): 4On-line CPU(s) list: 0-3Thread(s) per core: 1Core(s) per socket: 4Socket(s): 1NUMA node(s): 1Vendor ID: GenuineIntelCPU family: 6Model: 30Model name: Intel(R) Xeon(R) CPU X3430 @ 2.40GHzStepping: 5CPU MHz: 2394.116BogoMIPS: 4788.23Virtualization: VT-xL1d cache: 32KL1i cache: 32KL2 cache: 256KL3 cache: 8192KNUMA node0 CPU(s): 0-3*

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mod_verto Signaling and websocket closed
by martin.hoole＠emailn.de 16 Dec '16

16 Dec '16

Can you configure mod_verto to keep the RTC media connection even when the websockets is closed after signaling? I was looking for a REST API for signaling, and didn't find one for SDP Offer/Answer exchange. So I ended up using the websocket in mod_verto. Martin

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