July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call splitting by audio in/out
by Stanislav Sinyagin 15 Jul '13

15 Jul '13

It looks like it's not available in the dialplan, but at the C library level it seems to be doable: it will be great if a call could be split into two b-legs, one for input audio, and the other one for output. There are a number of usage scenarios when an operator needs to listen and to speak to two different groups of people. This can be partially achieved with a set of conferences bridged together with mute option, but something more straightforward would be great. Another interesting scenario is when the operator speaks two-way with someone, and is also able to listen to some other channel at the same time. regards, stanislav

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xml-curl-mod on call ANSWER
by adahary 15 Jul '13

15 Jul '13

I'm using xml-curl-mod to handle directory/dialplan events and xml-cdr-mod to handle CDRs. I'm missing the Dialplan XML POST that should be triggered when a call is answered. How can I get the an XML POST when call is ANSWERed ? Thanks Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/xml-curl-mod-on-call-ANSWER-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH adds WebRTC support to new 1.4 BETA.
by Shahrzad A. 15 Jul '13

15 Jul '13

Hi I'm also having the exact problem as Ashley! After upgrading to 'OpenSSL 1.0.1e 11 Feb 2013' I can not even successfully pass the 'make' part and I get "./.libs/libfreeswitch.so: undefined reference to `SSLv2_client_method" Any suggestion ? Sherry

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SIP attack SPA2102
by Miha 15 Jul '13

15 Jul '13

Hi, last night we noticed SIP attack from one of our costumar. Interesting thing is that the call come from Linksys SPA2012 which was behind NAT. The port forward was made for web access(xxx.xxx.xxx.xxx:8099). How it is possible do call from SPA2012 (relay calls)? I am looking at SPA but do not see any of that options. How SIP traffic could be send to SPA from which calls was send as there was no SIP port fwd on router? The device from which calls was send is SPA2012. THank you very much for all explenation!!! Miha

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originating an automated call, default value for origination_caller_id_number
by Rafal Gwizdala 14 Jul '13

14 Jul '13

Hi, Thanks for great support so far. I have another two questions to you: 1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. My originate command looks like this: originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/606999888 6668 The '6668' extension is an httapi script that plays an audio file and collects some digits. The problem is that my httapi ivr starts the playback even before the mobile phone is answerred. As a result, the called person doesn't hear the beginning of the recording. Can I somehow make my httapi script wait for the mobile to be answerred? 2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway. Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options? Thanks RG

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Building fscomm : can't find "stfu.h"
by arif hossain 14 Jul '13

14 Jul '13

Hi I'm trying to build fscomm. I'm on this commit : commit 7c88f35451b52cad104bb1794e2a2a2abef306c1 Author: Ken Rice <krice(a)freeswitch.org> Date: Mon May 6 18:30:47 2013 -0500 1.2.9 rev bump The error showed up : In file included from mainwindow.h:38:0, from main.cpp:32: ../src/include/switch.h:110:18: fatal error: stfu.h: No such file or directory compilation terminated. make: *** [main.o] Error 1 But : arif@khost:~/src/freeswitch/fscomm$ tree ../libs/stfu/ ../libs/stfu/ ├── CMakeLists.txt ├── stfu.c └── stfu.h Thanks in advance. ---------- Cheers Arif

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mod_sofia: Gateway variables not set for inbound calls
by Alexander Haugg 13 Jul '13

13 Jul '13

Hi, for my gateway in the sofia.conf.xml the variables will not set for an inbound call. for outbound it works successfully. here my profiles and gateway settings. is there a config fault? <configuration name="sofia.conf" description="sofia Endpoint">  <global_settings> <param name="log-level" value="0"/>  <param name="debug-presence" value="0"/> </global_settings> <profiles> <profile name="profile_H3K_42"> <param name="debug" value="0"/> <aliases> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="FreeSWITCH"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>  <param name="enable-3pcc" value="true"/>  <param name="inbound-use-callid-as-uuid" value="true"/>  <param name="outbound-use-uuid-as-callid" value="true"/> </settings> <gateways> <gateway name="H3K_42"> <param name="realm" value="${H3K_ip}:5060" /> <param name="username" value="" /> <param name="password" value="" /> <param name="from-user" value="" /> <param name="from-domain" value="${H3K_ip}" /> <param name="expire-seconds" value="600" /> <param name="register" value="false" /> <param name="retry-seconds" value="30" /> <param name="extension" value=""/> <param name="register-transport" value="udp"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="30" /> <variables> <variable name="prefix_pattern" value="^42(.+)$" /> </variables> </gateway> </gateways> </profile> </profiles> </configuration> Thanks for your Help!

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Re: [Freeswitch-users] RFC4575 and RFC4579 Freeswitch Version 1.5
by Jonathan Hunter 13 Jul '13

13 Jul '13

Hi Anthony, Thank you for replying. I have tried testing this with blink,Jitsi and Sipp script and seem to get same results each time. My configuration is basically terminate on 07917 which has; <action application="conference" data="$1@default++flags{rfc-4579}"/> However please see attached output of sofia global sip trace on.(RFC4575sofiasip). I can see Freeswitch responds with a 202 accepted to the SUBSCRIBE for event conference, then I see no NOTIFY responses. Any advice much appreciated as I want to ensure I haven't missed something in configuration. Many thanks Jon From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 85, Issue 123 To: freeswitch-users(a)lists.freeswitch.org Date: Sat, 13 Jul 2013 02:36:56 +0400 --Forwarded Message Attachment-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Fri, 12 Jul 2013 17:36:23 -0500 Subject: Re: [Freeswitch-users] RFC4575 and RFC4579 Freeswitch Version 1.5 I tested this and it does seem to work. Are you enabling sip trace on the console to see this sofia global siptrace on Also I had to update my Blink Lite for MAC up to the latest version in the app store. On Fri, Jul 12, 2013 at 7:24 AM, Jonathan Hunter <hunterj91(a)hotmail.com> wrote: Hi Guys, Hope you are all well? Im currently testing Freeswitch with a blink client to look to send events following RFC4575 /4579. I am making basic calls to a conference with flag rfc-4579 defined and the client connects to the conference, and FS as expected sends back the ;isfocus parameter on its contact address in the 200 ok. As a result the client I'm using (blink) sends a SUBSCRIBE, with Event: conference, which FS in turn sends a 202 Accepted response to. However after this point I get no NOTIFY messages at all to the initial SUBSCRIBE and for subsequent changes in the conference. I notice if a SUBSCRIBE for messages, or presence are sent, they are responded to without issue with a NOTIFY from Freeswitch. Please can someone advise if this is a configuration issue or something else that needs exploring? Any help would be great. Many thanks Jon _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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RFC4575 and RFC4579 Freeswitch Version 1.5
by Jonathan Hunter 13 Jul '13

13 Jul '13

Hi Guys, Hope you are all well? Im currently testing Freeswitch with a blink client to look to send events following RFC4575 /4579. I am making basic calls to a conference with flag rfc-4579 defined and the client connects to the conference, and FS as expected sends back the ;isfocus parameter on its contact address in the 200 ok. As a result the client I'm using (blink) sends a SUBSCRIBE, with Event: conference, which FS in turn sends a 202 Accepted response to. However after this point I get no NOTIFY messages at all to the initial SUBSCRIBE and for subsequent changes in the conference. I notice if a SUBSCRIBE for messages, or presence are sent, they are responded to without issue with a NOTIFY from Freeswitch. Please can someone advise if this is a configuration issue or something else that needs exploring? Any help would be great. Many thanks Jon

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Call redirect
by Shishko 12 Jul '13

12 Jul '13

Hi, when a user redirects its phone to mobile number, ITSP rejects call because caller id is not in our range of numbers. If I change CID to one of our numbers, call is accepted, but users would like to see original CID . Is there a way I could deflect a call to ITSP so user could see original caller id? Thanks, Denis

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