April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch and Freeradius integration
by Густаво Силва 03 Apr '15

03 Apr '15

Hi folks! Im getting some errors trying to enable AAA on my freeswitch with freeradius. I am getting the following errors: - When I start freeswitch: [ERR] mod_xml_radius.c:678 Failed to load radius handle for registration authentication - When I make a call: 2015-04-03 13:22:55.840618 [ERR] mod_xml_radius.c:566 Failed to load radius handle for digest invite authentication 2015-04-03 13:22:55.840618 [ERR] mod_xml_radius.c:879 Failed to create new accounting_start handle for call: 64b32ec2-d9eb-11e4-9c8c-85505819ce7f I can I solve this problem? My xml_radius.conf bellow: <configuration name="xml_radius.conf" description="Radius XML Gateway">  <auth_invite> <connection name="testing"> <param name="authserver" value="192.168.56.103:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/src/freeswitch-1.2.23/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="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param vendor="Cisco" name="h323-conf-id" variable="Core-UUID" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="ip" format="request-type=number"/> <param name="Called-Station-Id" variable="sip_to_user" format="%s"/> <param name="Calling-Station-Id" variable="sip_from_user" format="%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_invite> <auth_reg> <connection name="testing"> <param name="authserver" value="192.168.56.103:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/src/freeswitch-1.2.23/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> <auth_app> <connection name="testing"> <param name="authserver" value="192.168.56.103:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/src/freeswitch-1.2.23/src/mod/xml_int/mod_xml_radius/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_network_ip" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" format="dst-number-in=%s" /> <param name="Called-Station-Id" variable="sip_to_user" format="%s"/> <param name="Calling-Station-Id" variable="sip_from_user" format="%s"/> </fields> </auth_app> <auth_reg> <connection name="testing"> <param name="authserver" value="192.168.56.103:1812:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/share/freeradius/dictionary.cisco"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> </fields> </auth_reg> <acct_start> <connection name="testing"> <param name="acctserver" value="192.168.56.103:1813:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="0"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/src/freeswitch-1.2.23/src/mod/xml_int/mod_xml_radius/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-call-origin" variable="h323-call-origin" default="answer" format="%s"/> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_contact_host" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-out=%s" /> <param name="Calling-Station-Id" variable="sip_from_user" variable_secondary="ani" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_host" format="dst-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-number-out=%s" /> <param name="Called-Station-Id" variable="destination_number" format="%s"/> <param vendor="Cisco" name="h323-setup-time"/> </fields> </acct_start> <acct_end> <connection name="testing"> <param name="acctserver" value="192.168.56.103:1813:testing123"/> <param name="radius_timeout" value="10"/> <param name="radius_retries" value="0"/> <param name="radius_deadtime" value="0"/> <param name="dictionary" value="/usr/src/freeswitch-1.2.23/src/mod/xml_int/mod_xml_radius/dictionaries/dictionary"/> <param name="seqfile" value="/var/run/radius.seq"/> </connection> <fields> <param vendor="Cisco" name="h323-call-origin" variable="h323-call-origin" default="answer" format="%s"/> <param vendor="Cisco" name="h323-conf-id" variable_secondary="uuid" variable="originating_leg_uuid" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="uuid" format="h323-call-id=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_contact_host" format="src-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-in=%s" /> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_from_user" variable_secondary="ani" format="src-number-out=%s" /> <param name="Calling-Station-Id" variable="sip_from_user" variable_secondary="ani" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_host" format="dst-gw-ip=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" variable_secondary="dialed_extension" format="dst-gw-name=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="sip_to_user" variable_secondary="dialed_extension" format="dst-number-in=%s" /> <param name="Called-Station-Id" variable="destination_number" format="%s"/> <param vendor="Cisco" name="h323-setup-time"/> <param vendor="Cisco" name="h323-connect-time"/> <param vendor="Cisco" name="h323-disconnect-time"/> <param vendor="Cisco" name="h323-disconnect-cause"/> <param name="Acct-Session-Time" variable="billsec" format="%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable_secondary="progressmsec" variable="progress_mediamsec" format="pdd-time=%s"/> <param vendor="Cisco" name="Cisco-AVPair" variable="destination_number" format="dst-number-out=%s"/> </fields> </acct_end> </configuration>

3 11

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 03 Apr '15

03 Apr '15

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

1 0

WebRTC video conference
by Adam Ben-Ayoun 02 Apr '15

02 Apr '15

Hi guys, We are currently using Freeswitch as an MCU for audio only using WebRTC on the client. We want to add support for video (using VP8) such that in any point in time there is either no video or video is streaming from one member to all the others. Is there a way to achieve this using only Freeswitch? Thanks, Adam

4 8

Error Mod_lua.
by Antony Aguirre Morales 02 Apr '15

02 Apr '15

Hi, Install fusionpbx to manage freeswitch, perform the configuration of lua module enable these lines in lua.conf.xml <param name = "xml-handler-script" value = "app.lua xml_handler" /> <param name = "xml-handler-bindings" value = "dialplan | directory | configuration" /> I am using the script for the configuration that comes in the pbx in the following path: / var / www / html / fusionpbx / resources / install / scripts copy them to the following path freeswitch: / usr / local / freeswitch / scripts / but when you start freeswitch shows me the following errors have some idea: 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open /usr/local/freeswitch/scripts/resources/config.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:30: in main chunk 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute error! 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open /usr/local/freeswitch/scripts/resources/config.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:30: in main chunk 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute error! 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open /usr/local/freeswitch/scripts/resources/config.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:30: in main chunk 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute error! Regards.

3 3

the condtion fail when ${dialed_extension} =1001 but not fail when =1011
by Hagay Levi 02 Apr '15

02 Apr '15

I asked in freeswitch.org/jira in this link : https://freeswitch.org/jira/browse/FS-7335 , And Ítalo Ross <https://freeswitch.org/jira/secure/ViewProfile.jspa?name=italorossi> suggested me to send email to freeswitch-users mailing list. *Thanks in advance*

1 0

Happy April 1st
by Ken Rice 02 Apr '15

02 Apr '15

New Post on freeswitch.org from anthm check it out at http://ift.tt/1F4FlvZ Happy April 1st Spring is in the air and so is FreeSWITCH 1.6! Don’t forget to make plans for ClueCon 2015

3 2

Codecs (Again!)
by Andries Venter 02 Apr '15

02 Apr '15

I am using Freeswitch (1.4.8) as an SBC between our company PABX and our ISP's Trunk (also terminating on an SBC). My problem is that I need G729 in some cases and PCMA in others (the majority). The setup looks more or less like this: Our PABX (192.168.102.215)=>(192.168.102.102)Our SBC (10.17.159.10)=>(10.17.159.9)ISP SBC=>Rest of the world Since we use private addresses, Freeswitch must do RTP proxying. No transcoding must be done!!! I have set global_codec_prefs to "PCMA,G729". Then everything to destination supporting PCMA works. When trying to connect to a G729-only PABX, the call fails. Leg A is setup as follows 192.168.102.215.5060 > 192.168.102.102.5090: SIP, length: 933 INVITE sip:0873611337@192.168.102.102:5090;transport=udp SIP/2.0 To: <sip:0873611337@192.168.102.102;user=phone> From: "JGD Winson" <sip:anonymous@anonymous.invalid >;tag=snl_iFGEOrlLLf Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo CSeq: 1235 INVITE Contact: <sip:anonymous@192.168.102.215:5060;maddr=192.168.102.215> Via: SIP/2.0/UDP 192.168.102.215:5060 ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk Content-Type: application/sdp Content-Length: 288 X-Siemens-Call-Type: ST-insecure Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO Date: Wed, 01 Apr 2015 14:30:15 GMT Max-Forwards: 69 v=0 o=MxSIP 0 105140472 IN IP4 10.11.32.226 s=SIP Call c=IN IP4 10.11.32.226 t=0 0 m=audio 5014 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 396 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.102.215:5060 ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk From: "JGD Winson" <sip:anonymous@anonymous.invalid >;tag=snl_iFGEOrlLLf To: <sip:0873611337@192.168.102.102;user=phone> Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo CSeq: 1235 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit Content-Length: 0 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 1201 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.102.215:5060 ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-5nUh73XkOk From: "JGD Winson" <sip:anonymous@anonymous.invalid >;tag=snl_iFGEOrlLLf To: <sip:0873611337@192.168.102.102;user=phone>;tag=BytX6X1t7HytF Call-ID: SEC11-a64a8c0-1e64a8c0-1-9lz36fM4mXAo CSeq: 1235 INVITE Contact: <sip:0873611337@192.168.102.102:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit 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.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 226 Remote-Party-ID: "0873611337" <sip:0873611337@192.168.102.102 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1427881297 1427881298 IN IP4 192.168.102.102 s=FreeSWITCH c=IN IP4 192.168.102.102 t=0 0 m=audio 17318 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 When I change my global_codec_prefs to "G729,PCMA", G729 is offered to our PABX by Freeswitch, then to the ISP and then it works to destinations supporting G729. It fails to destinations not supporting G729. Now Leg A is setup as follows: 192.168.102.215.5060 > 192.168.102.102.5090: SIP, length: 934 INVITE sip:0873611337@192.168.102.102:5090;transport=udp SIP/2.0 To: <sip:0873611337@192.168.102.102;user=phone> From: "JGD Winson" <sip:anonymous@anonymous.invalid >;tag=snl_8NZMvB3KNA Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm CSeq: 1235 INVITE Contact: <sip:anonymous@192.168.102.215:5060;maddr=192.168.102.215> Via: SIP/2.0/UDP 192.168.102.215:5060 ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0 Content-Type: application/sdp Content-Length: 289 X-Siemens-Call-Type: ST-insecure Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO Date: Wed, 01 Apr 2015 14:27:51 GMT Max-Forwards: 69 v=0 o=MxSIP 0 1224193414 IN IP4 10.11.32.226 s=SIP Call c=IN IP4 10.11.32.226 t=0 0 m=audio 5014 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 396 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.102.215:5060 ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0 From: "JGD Winson" <sip:anonymous@anonymous.invalid >;tag=snl_8NZMvB3KNA To: <sip:0873611337@192.168.102.102;user=phone> Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm CSeq: 1235 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit Content-Length: 0 192.168.102.102.5090 > 192.168.102.215.5060: SIP, length: 1224 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.102.215:5060 ;branch=z9hG4bKSEC-a64a8c0-1e64a8c0-1-ZO003uQGh0 From: "JGD Winson" <sip:anonymous@anonymous.invalid >;tag=snl_8NZMvB3KNA To: <sip:0873611337@192.168.102.102;user=phone>;tag=g2eKXDUB6y4Hr Call-ID: SEC11-a64a8c0-1e64a8c0-1-k6bGtU73W2Vm CSeq: 1235 INVITE Contact: <sip:0873611337@192.168.102.102:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit 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.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "0873611337" <sip:0873611337@192.168.102.102 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1427866745 1427866746 IN IP4 192.168.102.102 s=FreeSWITCH c=IN IP4 192.168.102.102 t=0 0 m=audio 31726 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 I have tried various (hopefully all) combinations of inbound-codec-negotiation (generous, greedy and scrooge) and inbound-late-negotiation (true, false). Nothing does what I want. I believe Freeswitch needs to negotiate the codecs with the endpoint and choose one from my list. However, it seems to grab the first one in my list before checking the endpoint. Then it offers only that codec to the endpoint, which of course ignores / rejects it! What can I do? Regards Andries

2 3

Re: [Freeswitch-users] Transfer/shift calls from one freeswitch to other freeswitch box
by Luiz Fernando Softov 01 Apr '15

01 Apr '15

You can use gateway... Lets Say you have two Freeswitchs A 192.168.1.10 and B 192.168.2.10 In FS A you have a extension 1001, and in FS B extension 2001 In FS A, you need a gateway in external profile, to connect with FS B, its like: <include> <gateway name="gateway-fs-b"> <param name="proxy" value="192.168.2.10"/> <param name="register" value="true"/> <param name="username" value="2000"/> <param name="password" value="1234"/> </gateway> </include> You can read more in https://wiki.freeswitch.org/wiki/Clarification:gateways https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration Then when you make a call in FS A, in your dialplan, you need to make a bridge using gateway <include> <extension name="To FS B"> <condition field="destination_number" expression="(2...)"> <action application="bridge" data="sofia/gateway/gateway-fs-b/ ${destination_number}"/> </condition> </extension> </include> Now when you make a Call, to 2XXX, you shoud receive a call in FS-B In FS B, you need to have extension 2000, used by gateway to register and to make calls And of course, extension 2001. If you want to receive Caller ID 1001 of FS A dont use variables in extension 2000.xml <variable name="effective_caller_id_name" value="Extension 2000"/> <variable name="effective_caller_id_number" value="2000"/> It's a way to do this. -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

3 3

Re: [Freeswitch-users] Video frame capture
by Michael Jerris 01 Apr '15

01 Apr '15

The video support in 1.4 is pretty basic. 1.6 is going to add quite a bit of video support, you probably need to wait for that to do anything useful with video. > On Apr 1, 2015, at 5:40 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > > Hi guys, > > i'm trying to implement something like "preview" feature on video mailbox; just a simple picture of a frame of .fsv file. > Unfortunately i've no idea regarding .fsv file J > > Is it possible to "extract" an image from .fsv file? > > Any type of help is very appreciated J > > Many thanks > Marino Maria Mileti > marino.mileti(a)alice.it <mailto:marino.mileti@alice.it> > > <image001.jpg>Reduce your energy consumption and keep polar bears on ice! > > > > <http://www.avast.com/> > Questa e-mail è priva di virus e malware perché è attiva la protezione avast! Antivirus <http://www.avast.com/> . >

1 0

FreeTDM and OpenR2 2.x
by Charles Wang 01 Apr '15

01 Apr '15

Hi all, I want to build the freeswitch(v1.4.18) supported MFC-R2, And try to download the openr2-1.3.3.tar.gz. # tar zxvf openr2-1.3.3.tar.gz # cd openr2-1.3.3 # ./configure --prefix=/usr # make && make install # ldconfig After I configure freeswitch with command: ./configure --with-libpri --enable-debug I can find the message below: <<>> Modules and optional features checking for openr2_context_set_io_type in -lopenr2... no checking whether to build ftmod_r2... no checking for wat_version in -lwat... no checking whether to build ftmod_wat... no <<>> Digium libpri checking whether libpri is usable... yes checking whether libpri is too old... no checking whether libpri has BRI support... yes checking whether libpri has AOC event support... yes checking whether libpri has pri_maintenance_service()... yes <<>> Sangoma Wanpipe checking for sangoma_span_chan_toif in -lsangoma... no checking whether to build ftmod_wanpipe... no <<>> Sangoma ISDN stack checking for sng_isdn_init in -lsng_isdn... no checking whether to build ftmod_sangoma_isdn... no <<>> Sangoma SS7 stack checking for sng_isup_init_gen in -lsng_ss7... no checking whether to build ftmod_sangoma_ss7... no <<>> zlib checking for compress in -lz... yes checking whether zlib is installed... yes <<>> ftmod_misdn (Linux mISDN I/O plugin) checking for mISDN/mISDNif.h... no configure: mISDN/mISDNif.h not found <<>> Creating output files configure: creating ./config.status config.status: creating Makefile config.status: creating freetdm.pc config.status: creating mod_freetdm/Makefile config.status: executing depfiles commands config.status: executing libtool commands ============================ FreeTDM configuration ============================ + Modules Signalling: ftmod_analog....................... yes ftmod_analog_em.................... yes ftmod_isdn......................... no ftmod_libpri....................... yes ftmod_sangoma_isdn................. no ftmod_sangoma_ss7.................. no ftmod_r2........................... no ftmod_gsm.......................... no ftmod_pritap....................... no I/O: ftmod_zt........................... yes ftmod_wanpipe...................... no ftmod_misdn........................ no =============================================================================== In another word, is openr2 is not supported the freeswitch? After reading the README file, I found I should use openr2 2.X for freeswitch. But where to get the openr2 2.X ? -- Best Regards Charles

1 0