July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Client TLS certificate setup
by adahary 26 Dec '13

26 Dec '13

I have already installed the CA certificate in the FS conf/ssl (following FS/TLS wiki) and managed to get TLS registering with PhonerLite so I know that basically it should work with other sip/tls clients as well.I'm trying to setup other softphones with TLS/SIPS connection to Freeswitch and I'm failing to register with CSimpleSip/android and Jitsi/windows. I get this:tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb6e02268): new connection from tls/1.2.3.4:13886/sipstport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:962 tls_connect() tls_connect(0xb6e02268): TLS setup failed (error:00000001:lib(0):func(0):reason(1))tport.c:2092 tport_close() tport_close(0xb6e02268): tls/1.2.3.4:13886/sipsI will appriciate any help/instruction/tip on how to setup working TLS connection with CSimpleSip/android and Jitsi/windows clients.Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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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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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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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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Possible Polycom bridging issues in all recent FreeSWITCH versions
by Emrah 11 Oct '13

11 Oct '13

Hi, I have been experiencing the following issue since I upgraded my version of FS. In a scenario where a called is made from one Polycom VVX1500 to another, the video works both ways with only one way audio. A hold / unhold results in 2 way audio with 1 way video. I obviously suspected my configs, especially after subsequently trying the latest git, the 1.4 beta and the stable 1.2.10. So far, the only way I found to solve this issue is by using an old copy of FreeSWITCH with my current configs. FreeSWITCH Version 1.3.13b+git~20130119T155523Z~6cb3be7d8b (git 6cb3be7 2013-01-19 15:55:23Z) To clarify, the same config with any of the latest versions of FS does not work. Questions: How can I troubleshoot this further and help resolve this issue? Any one has an idea of what has happened? Thanks!

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Prevent A leg from hangup after bridge with inbound ESL socket
by Alex Massover 09 Oct '13

09 Oct '13

Hi, I have a very simple dialplan that just do park for incoming calls. All rest of leg management is done via ESL inbound socket. I'm trying to do the same behavior like in this dialplan example, but from ESL inbound socket: <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/internal/1001@$${domain}"/> <action application="bridge" data="sofia/internal/1002@$${domain}"/> The problem is with bridge API, if B leg doesn't answer (e.g. 404, or busy), A leg disconnects. But I'm trying to prevent A leg from disconnecting in order to do bridge to other place. Looks like hangup_after_bridge=false, park_after_bridge=true, transfer_after_bridge etc don't have any effect when bridge done from inbound socket. A leg disconnects always. Is there any way to keep A leg after bridge with inbound socket? I'm aware of originate, but prefer to user bridge. -- Best Regards, Alex Massover

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