September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Gateway Registration Issues
by Aloysius Lloyd 15 Dec '13

15 Dec '13

Hi All, I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have the following gateway configuration <include> <gateway name="wwc_test"> <param name="auth-username" value="username"/> <param name="username" value="14161231234" /> <param name="password" value="password" /> <param name="proxy" value="sbc1.iristel.net" /> <param name="realm" value="iristel.net" /> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> The above configuration is not working. I have the following error in cli Error. *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234 Registration Failed with status Conflict [409]. failure #1* ==== Also carrier saying there is conflict in the contact field.Here is how they receive now. *Contact: <sip:gw+test_iristel@10.20.30.1:5080 ;transport=udp;gw=test_iristel>.* But the contact field should be like below Contact: <14161231234@10.20.30.1:5080; Any suggestions? Thanks in advance. Lloyd

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JsSIP and wss
by Peter Steinbach 06 Dec '13

06 Dec '13

I am trying the Jssip demo with freeswitch and it works nicely with jssip 0.3.0 and the following configuration: ws_servers: 'ws://my_domain.com:8088' (My ws port is on 8088) However this only works with 'ws' but not with secure 'wss'. * In the log I can see that freeswitch sets wss-binding[:7443] * wss.pem is also present in the certs directory But with the config: ws_servers: 'wss://my_domain.com:7443' I do not even get a connection and I receive a 1006 error. Anybody has a working wss connection with Jssip and can share his config with me? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Re: [Freeswitch-users] mod_opus codec on Windows
by Jeff Leung 04 Dec '13

04 Dec '13

Perfect, I'll try it out when I have the time to do so. Thanks for going to places where I wouldn't have normally went through ;) Peter Olsson <peter(a)olssononline.se> wrote: I just wanted to notify that there is a patch available for this now (FS-5704). It has not yet been commited to git though. If anyone want to test it out, please apply the patch manually. /Peter 9 jul 2013 kl. 08:09 skrev Peter Olsson <peter(a)olssononline.se>: I think so too. I've started on a complete patch for the Windows build, hopefully I will have some time during this week to complete it. I'll submit this to Jira when finished. 2013/7/8 Michael Jerris <mike(a)jerris.com> > Are you sure symbol visibility is a problem even if these are in different > dll's ? I think this shouldn't be an issue. > > Mike > > On Jul 6, 2013, at 12:10 PM, Jeff Leung <jleung(a)v10networks.ca> wrote: > > The project was within FreeSWITCH, but the problem is because the way how > the opus lib builds, it includes silk and celt and I wasn’t going to have > FS to build all of that in tree because I was afraid the symbols from OPUS > would conflict with the symbols in CELT and SILK.**** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > users-bounces(a)lists.freeswitch.org] *On Behalf Of *Peter Olsson > *Sent:* Saturday, July 6, 2013 2:02 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] mod_opus codec on Windows**** > ** ** > Ah, ok. So you did the build of the lib outside fs, and then just pointed > to the built library?**** > ** ** > I'll start working to get this integrated in the Windows build, by > downloading the lib source files, unpacking them, and then building them.* > *** > ** ** > I will submit a Jira as soon as I have some progress. I might get back to > you with further questions.**** > ** ** > /Peter**** > > ** ** > 2013/7/6 Jeff Leung <jleung(a)v10networks.ca>**** > The project file is nothing more than having mod_opus to point to an out of > tree directory where the actual opus binary libs are built. > > The binaries I did was basically a bunch of hacks, there is quite some work > needed to be done with integrating the build process of opus with > FreeSWITCH's VS solution file > > > -----Original Message----- > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > > users-bounces(a)lists.freeswitch.org] On Behalf Of Peter Olsson > > Sent: Saturday, July 6, 2013 12:07 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_opus codec on Windows > > > > Good work! Could you also please file a Jira, and upload the project > files > and > > modifications that you've made for this to work? I'd be happy to assist > and > > get this commited. > > > > /Peter > > > > 6 jul 2013 kl. 08:20 skrev Jeff Leung <jleung(a)v10networks.ca>: > > > > > As I heard on last Wednesday's conference call, mod_opus isn't > "available" > > > on Windows, basically meaning that FreeSWITCH users can't fully > > > implement WebRTC on Windows Platforms just yet. > > > > > > I have taken the steps to have the codec successfully build on this > > > platform; However it's a pain in the neck to build as the opus library > > > generates multiple libs which includes CELT and SILK codecs (Go > figure). > > > > > > Shoot me a message off-list if you would like to use the binary I've > > > built for Win64 platforms. > > > > > > Before anybody asks for a 32-bit binary, the short answer is going to > be > no. > > > I won't be providing a 32-bit binary as I intend this binary to be > > > used in a production environment. > > > > > > > > > _________________________________________________________________________ > 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 > >

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FreeSWITCH not sending P-Asserted-Identity
by Vik Killa 29 Nov '13

29 Nov '13

I cannot get FS to send the P-Asserted-Identity header instead it always sends the P-Preferred-Identity header. According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I must set the privacy flag in order for P-Asserted-Identity to be used, so here is my dialplan entry and the SIP INVITE that is sent: <extension name="outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="lcr" data="$1"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="privacy=yes"/> <action application="privacy" data="yes"/> <action application="set" data="sip_h_Privacy=id"/> <action application="bridge" data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> </condition> </extension> ------------------------------------------------------------------------ INVITE sip:+17165555555@YYY.YYY.YYY.YYY:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp Max-Forwards: 68 From: "1000" <sip:7165551001@XX.XX.XX.XX>;tag=8U2rQ1pmXXaDQ To: <sip:+17165555555@YYY.YYY.YYY.YYY:5060> Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d CSeq: 27864844 INVITE Contact: <sip:mod_sofia@XX.XX.XX.XX:5060> User-Agent: FSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 209 X-FS-Support: update_display,send_info P-Preferred-Identity: "1000" <sip:7165551001@XX.XX.XX.XX> v=0 o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 17984 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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recommendations for Wifi SIP phones?
by Nathan Neulinger 26 Nov '13

26 Nov '13

Anyone got any recommendations for decent quality wifi sip phones? Obviously the 'android+soft client' model will work, but what about actual handsets built for the task. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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blind transfer back to caller on failure
by Tiago Sampaio 11 Nov '13

11 Nov '13

Hi all. I'm wondering if is possible to return the call if no answer or user busy? i tryed this: <context name="INTERNAL"> <extension name="INTER_RAMAL"> <condition field="destination_number" expression="^(\d{4})$"> <action application="set" data="continue_on_fail=true" /> <action application="bridge" data="[originate_timeout=10]user/$1" /> </condition> </extension> <extension name="BLIND_TRANSFER"> <condition field="destination_number" expression="^blind_transfer$"> <action application="answer"/> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 5000 #"/> <action application="set" data="continue_on_fail=true" /> <action application="execute_extension" data="is_transfer XML INTERNAL" />  <action application="bridge" data="user/${caller_id_number}" /> <action application="hangup" /> </condition> </extension> <extension name="is_transfer"> <condition field="destination_number" expression="^is_transfer$"/> <condition field="${digits}" expression="^(\d+)$"> <action application="set" data="continue_on_fail=true" /> <action application="transfer" data="-bleg ${digits} XML INTERNAL"/> <anti-action application="eval" data="cancel transfer"/> </condition> </extension> </context> Example: User A call user B, user B answer and do blind transfer to user C. User B hangup when call blind transfer and C rings 30 seconds and then transfer again to B before hangup A... Now if i do blind transfer and no answer or user busy, freeswitch hangup all sides...

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FS with SSL/TLS issues!
by mehroz 10 Nov '13

10 Nov '13

Hi, it seems like i am unable to get it done. Here is what i have done 1. created CA (Root) certificates: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org 2. created Server Certificate: ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org *I replaced pbx.freeswitch.org and freeswitch.org with my FS IP address in which i am listening.* 3. Enabled /internal_ssl_enable=true/ and/ ip_tls_version=tlsv1/ in vars.xml . 4. When i try to Register 3CX on TLS over port 5061, all i see is : tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0x7fbd94e7a5e0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2084 tport_close() tport_close(0x7fbd94e7a5e0): tls/59.103.76.82:56863/sips tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fbd94005090): events IN tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fbd94005090): new secondary tport 0x7fbd94f53c70 tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fbd94f53c70): new connection from tls/59.103.76.82:56864/sips. PLease help!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Is it necessary to call hangup....
by Moiz Chinoy 07 Nov '13

07 Nov '13

Hello, Is it necessary to call hangup in dialplan after say playing a file through playback application. -- Regards, Moiz Chinoy.

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enable-100rel question
by Mike Burlingame 24 Oct '13

24 Oct '13

I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? >From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files enable-100rel This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. <param name="enable-100rel" value="true"/>

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Video switching
by mehroz 23 Oct '13

23 Oct '13

Hi guys, I have Linphone SIP client and my UI allows me to switch call from audio to video and video to audio. After call establishment, when i switch to video, it do so ! and when i switch back to audio , it works. But after that when i switch to video , it doesn't! Any guess ? what could be wrong here? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-switching-tp7593439.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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